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01:14.09 | degenerate | I've got 100s of lines like this in my logs, what does it mean: |
01:14.09 | degenerate | [2018-01-09 20:10:38] WARNING[24434] pbx.c: Maximum PBX stack exceeded |
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01:18.44 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.5 (2017/12/22), Standard: 15.1.5 (2017/12/22); 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 |
01:21.31 | rmudgett | degenerate: Your dialplan is either nesting include contexts more than 128 levels or you have a circular context include arrangement. |
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03:54.19 | cj | moo |
03:55.07 | cj | I'm getting unicode characters echoed when I type things in to my asterisk console. anyone seen this before and maybe know how to fix it? |
03:55.19 | cj | Tim_Toady: you've always been able to help me with my unicode problems :-) |
03:55.42 | cj | wait, you're not TimToady, are you? |
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04:25.58 | elcontrastador | greetings guys...I'm getting a sip 'Forbidden' error call outbound thru my ITSP...inbound calling works fine. https://gist.github.com/elcontrastador/d1ad6e264a0fbeec5498f39262ea59aa |
04:35.33 | elcontrastador | I'm using Asterisk 15.1.5, btw |
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04:45.36 | elcontrastador | I have resolved the issue. My ITSP put had some fraud detection i needed to disable. |
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07:47.25 | bllackjack | Can anyone help me? I cannot find macro-stdexten in extensions. My ultimate goal is to get notified whenever a call happens |
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11:08.14 | velix | [TK]D-Fender: Thx. |
11:10.50 | velix | Can I record a phone call while it's running? I mean, start in the middle of the conversation? |
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12:25.28 | Haris | hello all |
12:27.23 | Haris | Is udp 5678 a significant port for voip/voice traffic ? |
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12:39.11 | Samot | Sure |
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12:45.28 | Haris | what does it do ? |
12:46.08 | Haris | its like traffic for this port is coming to us via default gw on route to telco 1 |
12:50.37 | velix | Sorry for asking in here, but I don't have an idea where to ask else. Does anyone know a service to check some hundret landline numbers for true existence? Sure, I could code it with Asterisk... but I'd love to use an existent service. |
12:57.17 | Samot | What do you mean "what does it do?" it's a port. |
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12:57.55 | Samot | velix: Not sure how Asterisk would do that properly. |
12:58.17 | Samot | velix: You need a number/line lookup service. |
12:58.21 | velix | Samot: Call a number and check the log? |
12:58.31 | Samot | OK |
12:58.32 | velix | Samot: Yeah, but 90% are for mobile only. |
12:58.39 | Samot | OK. |
12:58.47 | Samot | When you say "true existence" you mean? |
12:58.47 | velix | Samot: it's called a "ping" call. Like ring 1 sec and cancel the call. |
12:58.53 | Samot | It's a valid number format? |
12:58.59 | Samot | It's been assigned? |
12:59.03 | velix | Samot: Nah, that's easy. I mean, assigned. Yes. |
12:59.11 | Samot | Assigned does not mean routed. |
12:59.40 | Samot | Your ping call is going to what? Tell you something like "Number not in service?" |
12:59.44 | velix | Samot: Can't Asterisk show the debug info from the SIP trunk? |
12:59.46 | velix | Samot: yes. |
12:59.59 | Samot | Also, Asterisk doesn't handle SIP codes individually |
13:00.23 | velix | oh ok. Then I need to use Python and SIP API. |
13:00.34 | Samot | Making a "ping call" doesn't prove anything. |
13:01.04 | velix | Samot: in my country it does. We've got assigned or not assigned ;) |
13:02.09 | Haris | what voip/voice related traffic flows on udp port 5678? |
13:02.13 | Samot | Assigned to who |
13:02.22 | Samot | Haris: What ever you put on it. |
13:02.24 | Samot | It's a port. |
13:02.50 | Samot | I could use port 80 for SIP |
13:03.15 | Samot | Ports do not support "specific" traffic. |
13:03.18 | velix | Samot: I don't need the person's name. I just want to check, if it's assigned or "free" (unregistered). |
13:03.27 | Samot | To a person? |
13:05.32 | Haris | if there's nothing standard on it, then you just have to say you don't know |
13:05.47 | Haris | I already know if its something custom (I put something on it) |
13:06.00 | DanQuinney | velix: doing that in the UK would get you in very hot water with the regulator ;) |
13:06.02 | Haris | or nothing's on that port |
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13:06.32 | Haris | 18:06:23.180613 IP 172.31.3.58 > 10.20.80.2: ICMP 172.31.3.58 udp port sip unreachable, length 510 |
13:06.34 | velix | DanQuinney: I've checked it for Germany, it's legal here right now. |
13:07.29 | Samot | Haris: There is a list of "standard" ports available on the Internet. |
13:07.42 | Samot | That's about all it is though. |
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13:07.57 | Samot | Just like port 80 on TCP is "standard" for HTTP traffic |
13:08.00 | Haris | what does this mean ? it cant reach sip port ? |
13:08.06 | Samot | It doesn't mean you have to use it. |
13:08.13 | Samot | There's not such thing as a "SIP Port" |
13:08.19 | Samot | It's just a "port" |
13:08.26 | Samot | That listens on TCP and/or UDP |
13:08.37 | Samot | What you bind that to is up to you. |
13:09.03 | Samot | Again, you can use port 80 as a port for SIP |
13:09.09 | Samot | Instead of HTTP |
13:09.14 | Samot | It's up to you. |
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13:35.52 | flujan | hi there. Is it possible to exec a dialplan function inside a Agi Script? |
13:36.28 | Haris | what does the word unreachable signify in that tcpdump output ? |
13:36.32 | Haris | what was unreachable ? |
13:37.16 | Haris | and in what direction is something unreachable ? |
13:38.25 | Haris | 18:38:09.271894 IP 10.20.80.2.sip > 172.31.3.58.sip: SIP, length: 474 |
13:38.25 | Haris | 18:38:09.271912 IP 172.31.3.58 > 10.20.80.2: ICMP 172.31.3.58 udp port sip unreachable, length 510 |
13:39.25 | Haris | ah. my asterisk service is down. so its natural its going to say that |
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13:48.10 | [TK]D-Fender | flujan, yes |
13:55.42 | flujan | [TK]D-Fender: how so? Can you point me a example? |
13:55.53 | [TK]D-Fender | Geet it like any other variable |
13:55.58 | [TK]D-Fender | the idea is no different |
13:56.08 | [TK]D-Fender | or Set it |
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13:57.38 | Samot | Haris: Unreachable is pretty self explanatory. It can't reach the other side. In the case of SIP that means no response to a request was received. |
13:58.01 | Haris | yep. I just realized, this was a simple case |
13:58.28 | flujan | [TK]D-Fender: thanks. |
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14:46.39 | casix | hello |
14:47.25 | casix | how can I make a dial with pjsip with all information (proxy, user, password) in the cmd, without having any trunk? |
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14:53.33 | cervajs2 | outgoing call from pjsip webrtc endpoint with rtc-mux negotiate. pjsip endpoit has rtc-mux=yes . setRemoteDescription doesnt have a=rtcp-mux . is it correct? howto confirm that i'm using rtc-mux from asterisk side (and from browser)? thanks |
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14:57.45 | file | the option is rtcp_mux=yes and it would be "rtcp-mux" in the SDP |
14:58.00 | [TK]D-Fender | casix, You always have to use an endpoint. You can make it fairly empty though and pass the rest in-line |
15:00.09 | cervajs2 | sry. its ok on asterisk side "rtcp_mux : true" |
15:00.36 | casix | [TK]D-Fender, where can I find the syntax for the dial to pass the username and password in-line? |
15:03.18 | cervajs2 | file: invite from jssip is with a=rtcp-mux but response from asterisk is without a=rtcp-mux . its on ast 13.17.2 . do you think it can be bug? |
15:06.12 | file | it's possible but you'd really need to upgrade to the latest version first |
15:06.38 | file | I just confirmed in 15 it works as expected |
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15:12.03 | cervajs2 | file: how is rtcmux negotiate/require propagated to sip? |
15:12.15 | file | I don't understand the question |
15:12.37 | cervajs2 | i mean pc = new RTCPeerConnection({rtcpMuxPolicy: "negotiate"}) |
15:12.56 | file | it's not, that's policy within the browser over its behavior based on the result of the SDP negotiation |
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15:18.47 | cervajs2 | ops. learning time. rtcpMuxPolicy: "negotiate" is not clear for me. i'll back later |
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16:53.21 | Guest91029 | Have anyone seen this odd behaviour? It started when we moved from local database to Amazon RDS. Using Asterisk Realtime |
16:53.22 | Guest91029 | http://codepad.org/rvZpQU5L |
16:53.44 | Guest91029 | The Asterisk/CLI is frozen or does not respond. Also I did a tcpdump and can see no queries being made. This occurs randomly. |
16:55.32 | Guest91029 | Restart of Asterisk service does not work, either does restarting the server |
16:55.41 | file | you'd have to get a backtrace to show where it is blocked |
16:56.06 | file | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
16:57.18 | Guest91029 | Seems to work when I started with safe_asterisk all of a sudden |
16:58.55 | file | any disturbance with the database will make Asterisk upset |
16:59.51 | Guest91029 | No core dump was created |
16:59.55 | Guest91029 | Meaning it did not crash? |
17:00.36 | Guest91029 | Sure database disturbance I get that, I want to know why though so that I can fix it. |
17:01.14 | Guest91029 | It SEEMS, that doing a FLUSH HOSTS; in the mysql CLI sorts it out |
17:01.17 | Guest91029 | It does not make any sense |
17:01.43 | file | it didn't crash |
17:01.55 | file | it was likely in a deadlock in database code |
17:02.22 | Guest91029 | It worked excellent before we moved the database to RDS. I must say. |
17:03.45 | Guest91029 | Any suggestion on how to approach this issue? |
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17:13.49 | file | a backtrace would be needed to know precisely where it is blocked |
17:13.56 | file | the result of which would dictate what to look into |
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17:23.31 | Guest91029 | So asterisk needs to crash? Or is one created just by running safe_asterisk? |
17:23.51 | file | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock |
17:24.06 | file | that details how to get the info |
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17:49.04 | Guest91029 | @file: yeah ok |
17:49.08 | Guest91029 | So I have to recompile asterisk |
17:49.27 | Guest91029 | To bad I cannot replicate this on demand |
17:57.32 | Guest91029 | Bugger to be having to run asterisk with "seriously impact performance" for weeks until it might occur again :-( |
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18:32.50 | LunaLovegood | Is there a way to have a BLF dialplan hint for an extension number instead of for PJSIP devices? Like, I want a BLF light on the SIP phone at the reception desk to go red when someone from the outside dials into the IVR (before any of the office phones rings). |
18:33.50 | LunaLovegood | Like, calls coming from the ITSP to a specific number (there's also a fax number and I don't want BLF lights to turn on when we get faxes.) |
18:34.15 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_DEVICE_STATE |
18:34.22 | file | the primitives exist to do it. |
18:36.23 | LunaLovegood | So I can do a Set(DEVICE_STATE(Custom:lamp1)=BUSY) like in the example, but how do I know when to set it to NOT_INUSE? There may be many simultaneous calls. |
18:36.52 | file | you'd need to write logic to somehow do it, otherwise there is nothing built in that explicitly does what you want |
18:38.01 | [TK]D-Fender | file, is there a hook for GROUP()? |
18:38.04 | [TK]D-Fender | for the count |
18:38.14 | LunaLovegood | I'd need to deal with race conditions and other multithreading issues in the dialplan? |
18:38.23 | file | you can query the current count for a group |
18:39.35 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_GROUP_COUNT |
18:39.57 | file | unfortunately that's the best I've got |
18:40.54 | LunaLovegood | That should work, I think, thanks! |
18:41.51 | [TK]D-Fender | Group_count + hangup handlers could probably deal with this pretty efficiently |
18:42.09 | [TK]D-Fender | low liklihood of race |
18:51.09 | velix | [TK]D-Fender: Are you a rentable Asterisk consultant? |
18:52.23 | [TK]D-Fender | possibly. PM me |
19:02.37 | LunaLovegood | It does work! Thanks again! |
19:07.12 | LunaLovegood | I have to do more dialplan "programming" if I want to have the users be able to decide for themselves whether they want a password on their voicemail or not, right? |
19:09.09 | [TK]D-Fender | * will always ask for the VM password |
19:09.14 | [TK]D-Fender | I supposethey COULD leave it blank |
19:09.18 | [TK]D-Fender | but it will still ask |
19:09.31 | [TK]D-Fender | at which point you'd just hit # for instance I believe |
19:09.43 | [TK]D-Fender | if you want it to skip outright, then that's in how you call it |
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