| 00:02.03 | *** join/#asterisk ttbakiatwoam (~ttbakiatw@205.122.215.59) |
| 00:05.53 | ttbakiatwoam | Hello everyone, I have an issue with remote softphones not being able to transmit solid audio to my PBX, I get spots of dead noise It happens across multiple ISPs and I have a feeling its my firewall however, I have followed their specs for configuring SIP I have opened the ports on the firewall to the world I am not sure what to test next. |
| 00:06.55 | ttbakiatwoam | im loosing maybe 40% of the audio being transmitted to the PBX the rest comes through just fine. |
| 00:06.58 | WIMPy | Are you using the line for other traffic? |
| 00:07.22 | ttbakiatwoam | It talks to the SIP provider |
| 00:46.09 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 00:46.09 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 00:50.35 | lorsungcu | ttbakiatwoam: same deal, show a call |
| 00:50.37 | Samot | chris349: Sorry I was working on a failover issue. |
| 00:50.54 | Samot | chris349: How is it "unreliable"? |
| 00:56.00 | *** join/#asterisk ttbakiatwoam (~ttbakiatw@205.122.215.59) |
| 00:56.13 | drmessano | Dude |
| 00:56.27 | drmessano | You lose 40% of your audio |
| 00:56.34 | drmessano | You're timing out on IRC |
| 00:56.45 | drmessano | YOUR CONNECTION is BORKED |
| 00:56.57 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 00:56.57 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 00:57.22 | ttbakiatwoam | I disconnected my laptop without turning on wifi it has nothing to do with my ISP. |
| 00:57.30 | drmessano | If you think it's the router, replace it |
| 00:57.52 | drmessano | Ports do not cause 40% packet loss |
| 00:58.02 | drmessano | Misconfiguration doesn't |
| 00:58.14 | drmessano | What kind of firewall is this? |
| 00:58.44 | Samot | He just quit. |
| 00:58.54 | Samot | He was impossible in #freepbx |
| 00:59.07 | Samot | Couldn't even pastebin a debug properly. |
| 00:59.15 | Samot | It was always garbage... |
| 00:59.38 | drmessano | Lol |
| 00:59.54 | Samot | No, I mean literally. |
| 00:59.57 | drmessano | Oh |
| 00:59.59 | Samot | It looked like encoded text |
| 01:00.07 | Samot | You couldn't even tell what it was. |
| 01:00.13 | drmessano | Linksys router? |
| 01:00.20 | drmessano | Latest DD-WRT |
| 01:00.24 | Samot | +8aSNGasdjeas3835=320 |
| 01:00.29 | Samot | That's what it looked like. |
| 01:00.31 | *** join/#asterisk jkroon_ (~jkroon@165.16.203.103) |
| 01:00.35 | drmessano | SUPERMODDED WIFI |
| 01:00.35 | Samot | Just 100's of lines o fit. |
| 01:00.43 | drmessano | Lol |
| 01:00.59 | Samot | Oh but the first one.. |
| 01:01.17 | Samot | He posted 17000 lines of logs/debugs |
| 01:01.29 | Samot | Then when I said "what device is being used" |
| 01:01.44 | Samot | He was like "2345 starts on line 9445" |
| 01:01.51 | drmessano | Hahahah |
| 01:01.59 | Samot | 9400 lines of fscking garbage before it. |
| 01:03.54 | Samot | So.. |
| 01:04.29 | Samot | I ordered dinner. Everyone started to bitch "Why didn't you order this or that? Why is it all the stuff you like the most?!" |
| 01:04.50 | Samot | I just looked at them and said.... |
| 01:04.56 | chris349 | Samot, My firewall goes down, thats an issue I need to address sure, but when it comes back pjsip shows its registered to the provider, gets the invite, and rejects it with a no matching endpoint error. Also my phones stop registering. Both issues are resolved when I restart the asterisk service. Ive been using asterisk since before 1.0 and have never seen such an issue with chans_sip. Then I see posts from 1 year ago |
| 01:04.56 | chris349 | that describe exacly the issue I have today: https://community.freepbx.org/t/going-pjsip/33595 |
| 01:05.13 | Samot | "Fuck you. I got more birthday wishes from random strangers on the Internet today than anyone in this house" |
| 01:05.29 | Samot | "Which was ZERO" |
| 01:06.33 | Samot | chris349: So your "registration" with the provider isn't a "live" thing. |
| 01:06.39 | drmessano | Well I got in on the action as soon as FB told me |
| 01:06.52 | Samot | When you register they store your IP and port aka your "location" |
| 01:07.07 | Samot | And hold it.. |
| 01:07.37 | Samot | It will delete itself after it notices you're not responding to keepalives or it timesout. |
| 01:07.58 | Samot | When you bring your firewall back up what does a SIP debug show? |
| 01:08.07 | Samot | Do you see REGISTER attempts to the provider? |
| 01:08.54 | Samot | How long do the firewall issues last? |
| 01:09.33 | chris349 | Samot, I dont know what that means. Asterisk shows a valid registration to the provider but when the provider sends INVITE it is rejected due to No Matching Endpoint. After I restart the asterisk service the INVITE is accepted |
| 01:10.03 | Samot | OK, do you have a way to check if your registered with the provider? |
| 01:10.05 | Samot | On their side? |
| 01:11.23 | chris349 | If its not registered why would I get an INVITE? |
| 01:11.31 | *** join/#asterisk jkroon_ (~jkroon@uls-154-73-35-201.wall.uls.co.za) |
| 01:11.38 | Samot | Let me try one more time. |
| 01:11.45 | Samot | You REGISTER with your provider. |
| 01:11.48 | Samot | All is good. |
| 01:11.52 | Samot | That registration... |
| 01:12.09 | Samot | Is nothing more then you telling the provider "Send calls to this IP and this port." |
| 01:12.22 | Samot | If your trunk goes down.. |
| 01:12.32 | Samot | It doesn't "drop" the registration wth the provider. |
| 01:12.37 | WIMPy | Do the calls come from different hosts? |
| 01:13.10 | Samot | The provider will timeout your registration when you don't respond to keepalives. |
| 01:13.35 | Samot | You're basically going to have to recreate the issue and show a call that happens *after* the firewall comes back up. |
| 01:13.52 | Samot | So we can see what the SIP messages are and how they are being handled. |
| 01:14.27 | chris349 | I register with my provider. The firewall crashes at 2 AM and stops passing traffic, I reboot it at 10 AM, asterisk re-registers but then rejects INVITE until I restart the asterisk service |
| 01:14.51 | Samot | Need to see those rejections. |
| 01:15.00 | Samot | And why asterisk is rejecting. |
| 01:15.28 | Samot | This is not a PJSIP issue. |
| 01:15.34 | Samot | I mean it can be. |
| 01:15.50 | Samot | But this is not just solely PJSIP, I've seen this on Chan_SIP. |
| 01:16.08 | chris349 | The error I get is res_pjsip/pjsip_distributor.c No matching endpoint found |
| 01:16.16 | Samot | Show a whole call |
| 01:16.20 | Samot | I get that.. |
| 01:16.27 | Samot | Asterisk doesn't think the trunk is registered. |
| 01:16.38 | Samot | There for the endpoint isn't there. |
| 01:16.41 | WIMPy | remembers a discussion about not being able to survive an IP change with pjsip. That might be related. |
| 01:18.01 | Samot | Is it an IP change? |
| 01:18.06 | Samot | Does the IP change? |
| 01:18.43 | Samot | We actually haven't touched if the WAN IP is static or dynamic. |
| 01:19.32 | WIMPy | It shouldn't matter. But there was that story where it actually did. |
| 01:20.34 | chris349 | No, neither the WAN nor LAN IP address changes |
| 01:20.55 | chris349 | The IP is not static but it rarely changes |
| 01:21.55 | Samot | I would need to see Asterisk Re-REGISTER and then failed invites. |
| 01:22.05 | chris349 | But it could be if the firewall is down for an extended time period the LAN interface on the asterisk machine looses its DHCP address |
| 01:22.18 | Samot | So the call is being dumped at the trunk level, not at the device level? |
| 01:22.30 | Samot | Why would you have DHCP on your PBX? |
| 01:22.47 | igcewieling | chris349: have you tried just rebooting the nat router, instead of restarting Asterisk? If so that might point to the router. |
| 01:23.30 | Samot | You mean the firewall that dies? |
| 01:23.34 | Samot | And comes back up.. |
| 01:23.53 | chris349 | Im already reboot the router because it locks up. I have several other insstallations with DHCP and chan_sip that do not have this issue |
| 01:24.10 | igcewieling | ah, I'd not scrolled that far back. |
| 01:24.16 | Samot | Then make it a chan_sip trunk. |
| 01:24.21 | Samot | problem solved. |
| 01:25.25 | Samot | Either provide proof that this is a PJSIP issue or move your trunk to chan_sip so it works, like it always does. |
| 01:26.09 | Samot | Otherwise talking about how this could be happening and going on about how it always works with chan_sip is a waste. |
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| 05:23.38 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 05:26.10 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
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| 06:02.05 | *** join/#asterisk joobz (~joobz@CPE-121-214-35-111.lnse4.win.bigpond.net.au) |
| 06:02.25 | joobz | hi guys.. is there a way to forwrad a call through asterisk to another number, whilst retaining callerid? |
| 06:03.06 | joobz | i basically cant set callerid on the public network unless i verify each number (which i cant do).. just want to forwad the call to my mobile retaining callerid tho |
| 06:29.59 | *** join/#asterisk electroglue (~electrogl@unaffiliated/electroglue) |
| 06:31.09 | electroglue | I need a simple solution to call a number wait until it picks up and dial dtmf then hangup. The extension I created just seems to not work. Can someone help me? |
| 06:38.50 | Samot | Well first you need an IVR. |
| 06:39.22 | drmessano | Second |
| 06:39.26 | Samot | Or are you talking calling outbound and dialing DTMF? |
| 06:39.29 | drmessano | First you need Asterisk |
| 06:39.44 | drmessano | He wants to dial a number, send the DTMF, and hangup |
| 06:40.14 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial |
| 06:40.58 | drmessano | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_SendDTMF |
| 06:42.04 | Samot | Got some options. |
| 06:42.43 | Samot | This for a door buzzer or something like it? |
| 06:43.01 | electroglue | Samot: it's for a gate opener |
| 06:43.11 | electroglue | I have asterisk...no need for an ivr |
| 06:43.22 | Samot | I had your call direction backwards. |
| 06:43.27 | Samot | No IVR for this. |
| 06:43.45 | Samot | Look at the two links we posted. |
| 06:43.53 | electroglue | 989,1,Dial(Local/XXXXX@outbound-allroutes,,D(*16wwww5555wwwwww9)) |
| 06:44.11 | electroglue | So this is what I have. I've tried SendDTMF too |
| 06:44.12 | Samot | Yeah, no... |
| 06:44.26 | Samot | Why are you sending the call as an "outbound" call? |
| 06:45.18 | Samot | And don't use a local channel |
| 06:45.22 | Samot | you're call the device at the door |
| 06:45.22 | electroglue | Samot well what I was hoping was to dial a number that asterisk picks up and generates the call |
| 06:45.48 | Samot | Is the gate buzzer attached to an ATA or something other FXS gateway for SIP? |
| 06:45.56 | Samot | Does it register to Asterisk? |
| 06:46.06 | electroglue | No it does not. It's a pots line |
| 06:46.32 | electroglue | I have to dial into it then put in a code wait...then another...wait etc |
| 06:46.46 | Samot | Yes but you don't want a local channel |
| 06:47.01 | Samot | A local channel inverts the direction of the channel when it's answered. |
| 06:47.15 | electroglue | Ok, so it shouldl say "SIP/" |
| 06:47.24 | Samot | So when the buzzer answers, Asterisk turns it into an inbound channel. |
| 06:47.39 | Samot | Yes. You should dial out your SIP trunk like any other call. |
| 06:49.02 | electroglue | Hrm when I put in sip it just doesn't like that. It won't even let me dial out |
| 06:50.15 | Samot | You need to Dial() out the trunk. |
| 06:50.20 | Samot | Not through your outbound routes. |
| 06:54.03 | electroglue | I'm trying some things now Samot. You've got me on the right path |
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| 07:13.09 | electroglue | I have the call going out but the dtmf doesn't seem to be sending like it should |
| 07:33.22 | drmessano | ok |
| 07:47.11 | electroglue | http://pastebin.com/1GHUYtCH |
| 07:47.29 | electroglue | This is what I have but it's only dialing out and not doing the other steps |
| 07:50.32 | Samot | That's wrong. |
| 07:50.35 | drmessano | Yep |
| 07:50.42 | Samot | You are in the Dial() until it disconnects. |
| 07:52.00 | electroglue | Ok so what do I need to do? |
| 07:52.11 | Samot | Dialplan executes incoming calls. |
| 07:52.35 | Samot | So when you Dial() it's "outgoing" |
| 07:52.55 | Samot | Look at the Dial() options. |
| 07:53.03 | Samot | You can use D to send DTMF |
| 07:53.15 | electroglue | I had that previously... |
| 07:53.20 | Samot | U or M or something to fire off a macro on the channel. |
| 07:53.25 | Samot | Well we didn't see anything. |
| 07:53.42 | Samot | Without actually looking at a call, we can't tell you why it didn't work. |
| 07:53.52 | drmessano | or use originate |
| 07:54.20 | electroglue | 989,1,Dial(SIP/trunk/number,,D(*23wwww4555wwwww9)) |
| 07:54.30 | Samot | Showing us that isn't nothing. |
| 07:54.36 | Samot | Show us an actuall _call_ |
| 07:54.44 | Samot | asterisk -rvvvvvvvvvv |
| 07:54.46 | Samot | make the call |
| 07:54.48 | Samot | ~pb |
| 07:54.51 | infobot | [pastebin] a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 07:54.56 | Samot | ^^ put it there. |
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| 08:00.06 | drmessano | ...... |
| 08:04.31 | drmessano | 10 mins? |
| 08:04.32 | electroglue | So I got it. I thought it didn't work but it's night and I couldn't see the gate open |
| 08:04.41 | drmessano | ok |
| 08:05.08 | electroglue | So that little thing I did worked but only because you guys pointed me in the right direction |
| 08:05.42 | electroglue | I still need to figure out why my diaplans suck but I have a massive migraine and that'll wait for later |
| 08:08.14 | electroglue | thank you guys |
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| 16:48.51 | justone | hey yall |
| 16:49.47 | justone | i have a question about phpagi. In incredible pbx how can I load my php files such as MyHelloWorld.php |
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| 17:03.39 | igcewieling | ~freepbx |
| 17:03.42 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:03.55 | igcewieling | that applies to incredible pbx too. |
| 17:06.35 | Samot | Don't push that POS in to that room. |
| 17:10.02 | igcewieling | When running Asterisk, rather than a GUI, you'd edit extensions.conf and add it there as part of an extension. |
| 17:13.29 | justone | igcewieling: totally makes sense |
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| 23:30.19 | kalib | Hi guys. What would be the best option/strategy for this: I have four numbers, and I want that on a specific option (let's say 1) in my URA, my URA send this call to one of the four numbers I have, ramdomly (to one that is available/not in use. |
| 23:35.58 | kalib | for example: |
| 23:36.13 | kalib | exten => 1,1,Dial(here I would like to have 4 phone numbers) |
| 23:38.06 | dan_j | Hi. How can I get the ${QUEUESTATUS} if the caller hangs up prior to answering? |
| 23:55.41 | Samot | dan_j: To the queue answering or the agent? |
| 23:56.20 | Samot | kalib: You dial all four numbers with the dial() command. |
| 23:56.23 | dan_j | I think I've figured it out. Not sure why i was confused. I can just put it in the h extension that called Queue() |
| 23:56.34 | kalib | Samot: what you mean? |
| 23:56.36 | Samot | kailb: There are various formats in the sip.conf.sample file. |
| 23:57.47 | Samot | dan_j: I'm not sure it's available there. I'm not 100%. But if the channel enters h, the channel has been destroyed and all variables are gone. Not sure if that applies to the queue as well. |
| 23:58.13 | dan_j | Seems to work. ${ABANDONED} is YES |
| 23:58.40 | Samot | OK. |