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| 09:26.23 | dan_j | Does Asterisk 13 remove the warning regarding sip repeats and failure to receive a response? |
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| 11:59.02 | RomanDcoz | Hi ALl |
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| 12:46.30 | dan_j | Asterisk just sent bananas and was doing this. Any ideas what went wrong? |
| 12:46.34 | dan_j | https://www.irccloud.com/pastebin/satk2PVW/ |
| 12:47.11 | dan_j | A killall -9 and restart fixed it, but I need to know what happened so I can prevent it from happening again. |
| 12:48.37 | dan_j | s/sent bananas/went bananas |
| 12:51.30 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 12:52.49 | Samot | What do the logs says? |
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| 12:54.45 | dan_j | Which logs? The asterisk logs all look identical to the CLI output. |
| 12:55.44 | WIMPy | Did you try to debug it? |
| 12:56.00 | dan_j | I think it's some sort of DoS attack. |
| 12:56.15 | dan_j | None of my transports are called 'asterisk' |
| 12:56.20 | dan_j | Looking at the logs. It started like this |
| 12:56.27 | dan_j | <PROTECTED> |
| 12:56.34 | dan_j | ERROR[9853] res_pjsip.c: Unable to retrieve PJSIP transport 'asteritk' |
| 12:56.49 | dan_j | ERROR[10951] res_pjsip.c: Unable to retrieve PJSIP transport 'asteriuk' |
| 12:56.57 | dan_j | ERROR[20654] res_pjsip.c: Unable to retrieve PJSIP transport 'asteriwk' |
| 12:57.03 | dan_j | Actually. Maybe a memory leak? |
| 12:58.00 | dan_j | WIMPy: Unfortunately, I got about 50 calls shouting at me to get it back online, so I couldn't debug it. |
| 12:58.50 | dan_j | It was running fine for a week without any problem |
| 13:00.02 | *** join/#asterisk samwierema (~samwierem@095-097-255-066.static.chello.nl) |
| 13:00.12 | dan_j | Since the killall and restart, I'm now getting this error "Unable to apply outbound proxy on request to qualify contact sip:finefoods_202@X.X.X.X:5060' |
| 13:00.29 | *** join/#asterisk robinak (~quassel@unaffilated/robink) |
| 13:00.40 | samwierema | How do I know which of the SIP status codes is used when setting the HANGUPCAUSE to 21 (AST_CAUSE_CALL_REJECTED)? https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings |
| 13:06.03 | Samot | dan_j: Logs are historical. When did the "freak out" start getting logged.. |
| 13:06.12 | Samot | And what happened in the logs just before the freak out? |
| 13:07.33 | Samot | samwierema: It will be in the SIP message with the status and code. |
| 13:07.59 | Samot | samwierema; Outside of that there's nothing in Asterisk to get status codes from the message. |
| 13:08.42 | samwierema | Samot: yeah, sorry, maybe my question is: how does Asterisk pick which of the 4 it uses? Now it only seems to use 603. |
| 13:08.52 | Samot | It doesn't. |
| 13:09.01 | Samot | That's from the provider. |
| 13:09.15 | WIMPy | dan_j: Well the thing is: Either you temporarily fix it ASAP or you look what's going on and really fix it. |
| 13:09.41 | dan_j | Samot: the first thing that occured was the freakout. Nothing immediately before it. |
| 13:10.00 | Samot | And you looked at all your other logs.. |
| 13:10.08 | Samot | Around the same timestamp? |
| 13:12.22 | samwierema | Samot: http://pastebin.com/WasaWznN the ${CUSTOM_HANGUPCAUSE} can be set by the Stasis application, or it's 21 |
| 13:12.37 | samwierema | Then I would be the provider? |
| 13:12.48 | Samot | No. |
| 13:12.54 | Samot | Who are you sending the call to? |
| 13:13.14 | Samot | If this is a call over the PSTN then the error is from your provider. |
| 13:15.21 | samwierema | It can be a call over the PSTN, but that is determined by the outcome of the Stasis application. E.g. if a user is not found the Stasis application sets the ${CUSTOM_HANGUPCAUSE} and continues the dial plan, causing a hangup (and no call to the provider/PSTN). |
| 13:16.57 | Samot | So why are you using ISUP/ISDN hangup codes? |
| 13:17.20 | Samot | Who are you sending those codes back to? |
| 13:21.14 | samwierema | To the supplier that is sending the call. But, maybe I'm confused: can I just send a (e.g.) 404 directly with https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Hangup? |
| 13:21.18 | Samot | dan_j: Have you considered someone was trying to make calls through you? |
| 13:21.57 | Samot | Sure. |
| 13:22.14 | Samot | You can send HangUp(code) |
| 13:22.23 | dan_j | Samot: At the moment, I've only considered that someone was trying to knock my services offline. |
| 13:22.34 | dan_j | Was just on the phone. Looking at the other logs now. |
| 13:22.40 | WIMPy | samwierema: No. Hangup takes the Q.850 codes. What that makes for SIP is hardcoded somewhere. |
| 13:22.42 | Samot | transport= is the transport to use for the connection. |
| 13:24.06 | Samot | I have yet to see a DoS attack using PJSIP. |
| 13:25.24 | samwierema | WIMPy: right, that's what I'm looking for I think, which code maps to which SIP response code |
| 13:26.01 | WIMPy | samwierema: Looks like you already found out. |
| 13:27.58 | dan_j | Samot: None of my transports are called 'asterisk'. |
| 13:28.06 | dan_j | Or anything like asterisk |
| 13:28.19 | samwierema | WIMPy: this you mean https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings? How then, when sending AST_CAUSE_CALL_REJECTED, does it pick one of the 4 possible codes? I would like the 404 to be sent, but right now all I'm getting is the 603. |
| 13:28.50 | WIMPy | samwierema: No there will only be one mapping in the other direction. |
| 13:28.52 | Samot | dan_j: Exactly. The transport= settings had very specific options. |
| 13:29.13 | Samot | So therefore, anyone trying to use another type of "transport" in their requests will be rejected. |
| 13:29.18 | WIMPy | So you will get nothing but 603. |
| 13:29.42 | dan_j | Samot: Unable to retrieve PJSIP transport 'asterisk', indicates to me that asterisk is looking for a transport called 'asterisk' |
| 13:29.50 | dan_j | But I dont have any called 'asterisk' in my config. |
| 13:29.52 | Samot | Because someone is trying to USE it. |
| 13:29.55 | Samot | RIGHT |
| 13:30.37 | Samot | Quit focusing on the fact *you* dont have a transport called those things. |
| 13:31.36 | Samot | transport-udp, transport-tls, transport-upd-nat, transport-udp-ipv6 |
| 13:31.47 | Samot | ^^ THOSE are the options. |
| 13:32.15 | Samot | Or "" |
| 13:33.00 | Samot | Someone sent a request to you, PJSIP says they are trying to use a "transport=asterik" <<- which doesn't exist. So yeah, it's going to reject it. |
| 13:33.43 | dan_j | Ok. So assuming someone is trying to use a transport called 'asterisk', what would be the point of then sending another request to use transport called 'asteriâk' |
| 13:33.51 | dan_j | apart from trying to knock out the system. |
| 13:34.08 | Samot | You didn't notice each attempt was a different spelling? |
| 13:34.10 | samwierema | WIMPy: Samot: ah, found it in the source code (hangup_cause2sip). Thanks for all the help! |
| 13:34.30 | Samot | samwierema: So what does that mean? That's a function in the source code. |
| 13:34.41 | dan_j | Samot: Yes, it was the 'S' digit incremented by 1 until it reached a non ascii character |
| 13:34.49 | Samot | Right. |
| 13:34.56 | samwierema | Samot: that I found the mapping, and what cause sends what |
| 13:35.08 | Samot | s is generally where you send bad external calls with your "default" context". |
| 13:35.48 | dan_j | And most of the IP addresses shown in the logs requesting that transport are from known trusted ips. So unless there is a way for a hacker to get all my endpoints to request an invalid transport, I believe this is an asterisk issue. |
| 13:36.03 | Samot | samwierema: You may not get the SIP codes you think on the otherside. |
| 13:36.24 | samwierema | Samot: how come? |
| 13:36.53 | Samot | samwierema: ISDN code 21 has FOUR SIP codes. |
| 13:37.23 | WIMPy | Samot: You do realize that that was the original question? |
| 13:37.35 | Samot | Which means anyone processing ISDN codes would take one of those 4 SIP codes it receives and "convert" it to the ISDN code. |
| 13:37.59 | WIMPy | Anyone? No. Asterisk. |
| 13:38.33 | Samot | Yes, I do. |
| 13:38.58 | Samot | But him sending a HangUp(21) doesn't mean it's going to be the exact SIP code on the other side. |
| 13:39.03 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:39.40 | WIMPy | What other side? It's about Asterisk. |
| 13:39.56 | dan_j | Samot: It's a bit of a stretch to say that the S in 'asterisk' is where invalid calls go. I think this is a memory leak. There must be something in asterisk that is incrementing the memory block containing the 's' of 'asterisk' by one. |
| 13:40.26 | Samot | dan_j: Do you have a context you send "unwanted" attempts to? |
| 13:40.35 | Samot | When someone makes an inbound request.. |
| 13:40.39 | dan_j | Yes. Default. and it's empty. |
| 13:41.04 | Samot | Well generally people put diaplan there to you know, log and handle those bad calls. |
| 13:41.09 | dan_j | Inbound requests from endpoints which are trusted (ie they are set up in the config using an IP) go to [incoming] |
| 13:41.37 | dan_j | Looking at the verbose logs, there were no bad calls at the time of the freakout. |
| 13:41.44 | Samot | So when people use the [default] context for unwanted calls, they use the s exten to handle it. |
| 13:41.47 | Samot | OK. |
| 13:42.26 | WIMPy | Are you sure you know what the s extension is? |
| 13:42.30 | dan_j | And I dont accept invites unless they have SIP auth credentials or from some specific IP addresses. |
| 13:43.01 | Samot | So you still let those people in.. |
| 13:43.08 | dan_j | WIMPy: I know what the 's' extension is, but I dont see the connection with the freakout I posted. |
| 13:43.19 | [TK]D-Fender | you should never have a context named [default] |
| 13:43.27 | dan_j | [TK]D-Fender: I dont. |
| 13:43.29 | [TK]D-Fender | and "s" is not where invalid calls go. |
| 13:43.34 | Samot | No, it should be changed. |
| 13:43.42 | Samot | I didn't say s is where invalid calls go. |
| 13:43.52 | [TK]D-Fender | Samot, wasn't correcting you |
| 13:44.18 | [TK]D-Fender | (this time, just wait, it'll come :p) |
| 13:44.19 | dan_j | [TK]D-Fender: What are your thoughts on this freakout? The server was running fine for weeks and suddenly went nuts. |
| 13:44.23 | dan_j | https://www.irccloud.com/pastebin/keve7USJ/ |
| 13:44.33 | WIMPy | dan_j: That question was for Samot. |
| 13:44.49 | [TK]D-Fender | ERROR[18159]: res_pjsip.c:2735 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'asteriâk |
| 13:44.58 | [TK]D-Fender | that looks like a messed up char in there |
| 13:45.13 | [TK]D-Fender | "asteriâk" |
| 13:46.19 | dan_j | Yep, but it started off with this |
| 13:46.26 | dan_j | Unable to retrieve PJSIP transport 'asterisk' |
| 13:46.34 | dan_j | then two seconds later |
| 13:46.34 | dan_j | Unable to retrieve PJSIP transport 'asteritk' |
| 13:46.48 | dan_j | I personally think it's a memory leak incrementing that character |
| 13:47.36 | Samot | WIMPy: Yes, its the "start" extension and yes it's not a catch all. Again, I didn't say it was a CATCH ALL I said people USE IT in their "bad request" contexts. |
| 13:47.38 | [TK]D-Fender | What are version you running? |
| 13:48.47 | [TK]D-Fender | Samot, "s" has no impact on calls that aren't typically from DAHDI. No general call should hit it otherwise |
| 13:49.00 | [TK]D-Fender | (knew that wouldn't take long!) |
| 13:49.10 | dan_j | [TK]D-Fender: v13.11.2 |
| 13:49.15 | Samot | OK. |
| 13:49.22 | *** join/#asterisk sotoz (~textual@095-097-255-066.static.chello.nl) |
| 13:50.06 | Samot | s has no place in non-DADHI calls. Got it. |
| 13:50.10 | dan_j | Although I've got another server running that without a problem. |
| 13:50.26 | [TK]D-Fender | dan_j, well you are more than a release behind... upgrade, retest... |
| 13:50.38 | WIMPy | There is no such thing as a start extension. Even if there seems to be some tradition to use s for starting macros. |
| 13:50.48 | dan_j | [TK]D-Fender: Will do. |
| 13:51.06 | [TK]D-Fender | <Samot> s has no place in non-DADHI calls. Got it. <- not "no" place, but no RANDOM call should target it. It doesn't catch anything extept calls explicitly targeting tthe letter |
| 13:51.31 | WIMPy | Why would s be dahdi specific? It will also be used by SIP. |
| 13:51.38 | [TK]D-Fender | WIMPy, DAHDI uses it, as doees another "meh" channel driver or two |
| 13:51.50 | [TK]D-Fender | WIMPy, Analog has no DID, that's why |
| 13:52.25 | [TK]D-Fender | WIMPy, And VoIP-wise if you register and set nothing to target at the end, Asterisk SENDS them "s" as the target |
| 13:52.36 | [TK]D-Fender | Seems no-one really remembers what "s" is for... |
| 13:52.49 | [TK]D-Fender | ~stdextens |
| 13:52.49 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 13:52.52 | [TK]D-Fender | ^ |
| 13:53.14 | WIMPy | Well, s is a replacement for "" (nothing). |
| 13:53.59 | [TK]D-Fender | philosophically I suppose in a way... |
| 13:55.54 | WIMPy | Certainly much discussed and leading to a bit of confusion over and over again. |
| 13:56.12 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-jxpbmgxngpiaehpc) |
| 13:56.12 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:57.52 | Samot | This is part of the reason I haven't moved to Chan_PJSIP yet. |
| 13:58.47 | Samot | Because I only allow domains with Chan_SIP. |
| 13:59.21 | Samot | and the IP of the PBX is not an allowed domain, so requests sent to the IP are just rejected. |
| 14:00.03 | WIMPy | Yes. That halps a lot. |
| 14:00.40 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:01.01 | Samot | Well that and requests have to go through the Kamailio server... |
| 14:04.55 | Samot | Next step is to finally setup the private network on the systems at my cloud DC. |
| 14:06.10 | Samot | So the Asterisk boxes aren't listening on public IPs anymore and using Kamailio to handle anything public facing. |
| 14:10.23 | Samot | The combination of decent IPTables, SIP domains and Kamailio has taken a lot the BS load of the Asterisk and FreePBX servers. |
| 14:14.17 | *** join/#asterisk metanovii (~metanovii@176.49.170.21) |
| 14:14.24 | metanovii | hi dudes |
| 14:14.26 | metanovii | \o |
| 14:17.05 | dan_j | [TK]D-Fender: ever since the freakout, I'm getting this and peers can't connect. Any ideas? |
| 14:17.16 | dan_j | Unable to apply outbound proxy on request to qualify contact sip:company_201@217.155.XXX.XXX:38468 |
| 14:17.21 | metanovii | asterisk 13 have opus codec? |
| 14:17.25 | dan_j | Only some can't connect. Others are fine. |
| 14:18.07 | dan_j | [TK]D-Fender: I need to flush out those contacts out of the asterisk internet database I think. |
| 14:18.20 | [TK]D-Fender | metanovii, https://www.google.ca/#q=asterisk+13+opus |
| 14:24.54 | *** join/#asterisk nohitall (~nohitall@unaffiliated/nohitall) |
| 14:26.31 | nohitall | hi ugys |
| 14:27.13 | nohitall | Having some weird issues in the office, 2 out of 4 phones suddenly fail to register, I only see SIP/2.0 401 Unauthorized in the logs, just wonder how to debug this |
| 14:27.26 | nohitall | nothing changed in configs |
| 14:28.48 | [TK]D-Fender | show the actual full debug |
| 14:29.33 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 14:31.09 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 14:34.36 | *** join/#asterisk kahlb_ (~kahlb@dslb-146-060-147-119.146.060.pools.vodafone-ip.de) |
| 14:36.56 | dan_j | I'm trying to delete some data from astdb because i'm getting loads of these. 'Unable to apply outbound proxy on request to qualify contact' |
| 14:37.10 | dan_j | But whenever I try to delete, it says it's deleted but it doesnt actually get deleted. |
| 14:37.27 | dan_j | Identical issue as this post http://forums.asterisk.org/viewtopic.php?f=1&t=92546 |
| 14:45.17 | *** join/#asterisk Deeewayne (~dwayne@c-68-32-120-188.hsd1.al.comcast.net) |
| 14:45.55 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 14:46.46 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 14:48.12 | dan_j | Where is astdb stored? I think I just need to delete it as it's not allowing me to delete keys from within asterisk |
| 14:48.54 | dan_j | Never mind. I've found it. |
| 14:51.17 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 14:53.37 | *** join/#asterisk Mitesh (1b6a65ec@gateway/web/freenode/ip.27.106.101.236) |
| 14:54.10 | Mitesh | hello, I am having a question for application Queue |
| 14:54.50 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 14:55.10 | Mitesh | a call going to queue and then found one agent to call. Call is connected with agent, but still the first call is not comming out from Queue application |
| 14:55.39 | Mitesh | is this normal flow? shouln't it come out from app_queue? |
| 14:56.18 | Mitesh | I mean the first channel is not coming out from app_queue |
| 14:58.51 | dan_j | Mitesh: Look at 'core show application Queue' |
| 14:59.02 | dan_j | Option c: Continue in the dialplan if the callee hangs up |
| 14:59.55 | [TK]D-Fender | Show us the call |
| 15:00.50 | Mitesh | Well callee is not hangup up |
| 15:00.57 | Mitesh | both are in the call only |
| 15:01.21 | [TK]D-Fender | Show us the call |
| 15:04.57 | Mitesh | kindly find consol log here http://pastebin.com/b0Vmm0aN |
| 15:06.17 | dan_j | Once in the queue and the call is answered, there is no more dialplan processing. |
| 15:06.20 | [TK]D-Fender | So queue is calling a local channel, and THAT is doing its own dial |
| 15:06.20 | WIMPy | The Queue application bridges the call, or what is your exact question? |
| 15:06.22 | dan_j | Just like Dial when someone answers. |
| 15:06.46 | [TK]D-Fender | And indeed Queue() is still processing the call, just like Dial() |
| 15:07.14 | dan_j | WIMPy: I think he wants to execute more dialplan code once the call is answered |
| 15:07.40 | Mitesh | Yes, |
| 15:07.45 | Mitesh | basically, |
| 15:08.03 | *** join/#asterisk kharwell (kharwell@nat/digium/x-yhwoetuqcrqiowxh) |
| 15:08.03 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:08.05 | [TK]D-Fender | If he wants something justt becfore the bridge he can either use a macro/gosub in the Queue() call, or the Dial() |
| 15:08.17 | [TK]D-Fender | But there is nothing "during" the bridged call |
| 15:08.19 | Mitesh | I want that the channel come out from the app_queue and then I can do something more on that |
| 15:08.20 | [TK]D-Fender | that does not happen |
| 15:09.03 | Mitesh | can i say app_queue is a blocking application just like dial? |
| 15:09.10 | *** part/#asterisk nohitall (~nohitall@unaffiliated/nohitall) |
| 15:09.13 | WIMPy | yes |
| 15:09.40 | WIMPy | Actually most applications are blocking. |
| 15:09.50 | Mitesh | It means that the channel will come out only after the hangup of that bridge? |
| 15:10.06 | WIMPy | yes |
| 15:10.08 | Mitesh | i.e. caller or callee will hangup the call |
| 15:10.13 | Mitesh | ah okay |
| 15:10.25 | dan_j | Mitesh: What do you want to do after app_queue? |
| 15:10.37 | [TK]D-Fender | I just gave you 2 ways to call your code just before they are actually talking |
| 15:10.40 | WIMPy | Or if the queue is ful or the maximum waiting time was exceeded. |
| 15:10.45 | WIMPy | full |
| 15:11.08 | Mitesh | okay |
| 15:11.47 | WIMPy | Hmm. Or if you have configured the possibility to exit the queue via DTMF. |
| 15:12.32 | Mitesh | how can i do it with DTMF? |
| 15:13.03 | Mitesh | will it distory the call or call will continue |
| 15:13.11 | Mitesh | I want that call will continue |
| 15:13.13 | WIMPy | Chech the sample config. It's a feature of app_queue. |
| 15:13.49 | [TK]D-Fender | Careful on WHEN you can do that |
| 15:14.16 | [TK]D-Fender | Mitesh, I jsut GAVE you the only 2 things you can do before they talk. |
| 15:16.45 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 15:17.13 | Mitesh | well, basically using local channel I am dialing an extension which is on another machine another instance of the asterisk |
| 15:17.58 | Mitesh | after the extension got connected, I want that the caller also do not stay in app_queue and move to that another machine |
| 15:18.34 | *** join/#asterisk skyroveRR (~skyroveRR@unaffiliated/skyroverr) |
| 15:18.38 | Mitesh | so both does communication the another server and this server will become free from SIP channel |
| 15:18.56 | Mitesh | after processing queue feature |
| 15:19.01 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-bntpvoseynjmlyzo) |
| 15:19.11 | Mitesh | i mean got free after processing queue feature |
| 15:23.20 | Mitesh | can you suggest any good way to achive this? |
| 15:23.43 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:1999:7f27:6c5f:2e68) |
| 15:26.22 | [TK]D-Fender | that is not a normal feature at all |
| 15:26.35 | Mitesh | yes, that is not normal |
| 15:26.58 | Mitesh | but can that be achived? |
| 15:27.12 | [TK]D-Fender | If you are expecting them to leave the server then you'll have to make a few assumptions : first is thatt both ends will allow re-invites, second is making sure that * has no reason to stay in the patth |
| 15:27.32 | [TK]D-Fender | Where is the FIRST leg of the call coming from exactly? |
| 15:27.41 | [TK]D-Fender | Where are you calling out out to? |
| 15:27.48 | Mitesh | yes |
| 15:28.29 | Mitesh | first leg is at the another server where i am calling out the local channel |
| 15:29.07 | [TK]D-Fender | WHERE is it |
| 15:29.13 | [TK]D-Fender | what NETWORKS |
| 15:29.17 | [TK]D-Fender | for both legs |
| 15:29.27 | Mitesh | in the local network |
| 15:29.43 | [TK]D-Fender | then set both of your peers to allow reinvites. |
| 15:29.54 | Mitesh | okay... |
| 15:30.09 | [TK]D-Fender | They should then send media directly, but that will not stop the SIGNALLING from staying with * |
| 15:30.28 | Mitesh | okay |
| 15:31.22 | Mitesh | can I not move signalling from this *? |
| 15:31.45 | Mitesh | and move completely to the another * |
| 15:31.46 | Mitesh | ? |
| 15:32.22 | [TK]D-Fender | no |
| 15:32.24 | Mitesh | i mean move signalling where the media is moving |
| 15:32.27 | [TK]D-Fender | not with Queue in the way |
| 15:32.32 | Mitesh | i see |
| 15:33.32 | skyroveRR | I'm running asterisk 13.9.1 on my custom musl linux distro with the chan_mobile module loaded. My dial plan basically answers the call, waits for 10 seconds for any DTMF input and hangs up with the "vm-goodbye" file, however not a single sound file is being played, including the vm-goodbye.gsm file. Any ideas? |
| 15:34.12 | Mitesh | and leg will comeout from Queue only when any leg hangup |
| 15:34.34 | [TK]D-Fender | skyroveRR, idea : SHOW US |
| 15:34.41 | Mitesh | so basically before call hangup or timeout or queue threshold there is no way that leg will comeout, right? |
| 15:34.57 | [TK]D-Fender | huh? |
| 15:35.03 | [TK]D-Fender | it STAYS in queue |
| 15:35.04 | Mitesh | comeout from queue, right> |
| 15:35.07 | Mitesh | right? |
| 15:35.14 | Mitesh | okay |
| 15:35.17 | [TK]D-Fender | the MEDIA might be direct if you keep out of the way and are lucky |
| 15:35.31 | [TK]D-Fender | but that remains a call in progress |
| 15:35.40 | skyroveRR | [TK]D-Fender: http://pktsurf.in/files/asterisk.log.txt |
| 15:35.43 | [TK]D-Fender | it does NOT exit until the call has ended |
| 15:36.05 | Mitesh | Okay |
| 15:36.23 | Mitesh | thank you, thank you very much for help |
| 15:37.13 | [TK]D-Fender | skyroveRR, Looking like an issue with chan_mobile directly... |
| 15:39.23 | skyroveRR | Well, the paired phone has got full volume set and so has the other phone that's making the call. |
| 15:40.23 | skyroveRR | If asterisk was unable to play vm-goodbye.gsm, would it result in an error and would it have shown up in the command line? |
| 15:40.31 | [TK]D-Fender | not sure what else can go wrong on that driver... |
| 15:41.08 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 15:41.23 | skyroveRR | The paired phone can dial local numbers and output the audio well.. |
| 15:41.32 | skyroveRR | Anything got to do with ALSA on the distro? |
| 15:45.57 | [TK]D-Fender | no idea |
| 15:49.53 | skyroveRR | [TK]D-Fender: which conf files within /etc/asterisk are responsible for basic audio playback? |
| 15:54.36 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-14.wall.uls.co.za) |
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| 15:59.40 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 16:05.00 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
| 16:05.58 | [TK]D-Fender | No such thing |
| 16:06.35 | skyroveRR | :| |
| 16:08.19 | *** join/#asterisk hyegeek (~hakimian@wsip-72-214-228-246.ph.ph.cox.net) |
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| 16:13.59 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:19.05 | dan_j | What is the pjsip equivalent of 'sip show peer XXX'? I want to see the useragent of an endpoint |
| 16:22.32 | *** join/#asterisk miralin (~Thunderbi@194.8.128.50) |
| 16:28.52 | *** join/#asterisk areski (~areski@15.red-83-33-189.dynamicip.rima-tde.net) |
| 16:29.46 | igcewieling | 24 hours with no activity on #asterisk. |
| 16:39.12 | *** join/#asterisk areski (~areski@15.red-83-33-189.dynamicip.rima-tde.net) |
| 16:43.46 | *** join/#asterisk quentusrex (~quentusre@freeswitch/developer/quentusrex) |
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| 16:59.52 | yahoosrs | hello, anyone available to assist? I have FreePBX installed running Asterisk 11.2, Earthlink provided SIP trunk, I can't get outgoing/incoming to work. |
| 17:00.07 | *** join/#asterisk overyander (~Jeff@12.49.160.131) |
| 17:00.52 | yahoosrs | I have two phones up right now to test, they can ring each other. I can ping the Adtran, earthlinks device. Guy tells me I have to set up a route. I don't understand what he's talking about. |
| 17:01.01 | *** join/#asterisk eric_hill (~eric_hill@wsip-184-180-163-60.ks.ks.cox.net) |
| 17:05.09 | [TK]D-Fender | igcewieling, Who? |
| 17:05.38 | [TK]D-Fender | yahoosrs, |
| 17:05.40 | [TK]D-Fender | #freepbx |
| 17:05.48 | [TK]D-Fender | ~#freepbx |
| 17:05.50 | *** join/#asterisk mknooihuisen (~mknooihui@12.150.48.70) |
| 17:05.52 | [TK]D-Fender | ~freepbx |
| 17:05.52 | 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 |
| 17:05.54 | [TK]D-Fender | ^^^ |
| 17:06.08 | igcewieling | yesterday: (11:19:05 AM) dan_j: What is the pjsip equivalent of 'sip show peer XXX'? I want to see the useragent of an endpoint |
| 17:07.03 | [TK]D-Fender | dan_j, pjsip <TAB> |
| 17:07.11 | [TK]D-Fender | * CLI has AUTOCOMPLETE. Use it |
| 17:07.27 | dan_j | Tried all of the available options but cant see a similar output. |
| 17:07.29 | [TK]D-Fender | This isn't something that should have taken more than a minute without having to ask at all... |
| 17:07.39 | [TK]D-Fender | it isn't taht similar |
| 17:07.42 | [TK]D-Fender | ther is no perfect parity |
| 17:07.50 | [TK]D-Fender | There is a clear one for ENPOINTS & AOR |
| 17:07.58 | [TK]D-Fender | What you get is what you get |
| 17:08.14 | [TK]D-Fender | there is nothing more buried outside that "pjsip" keyword |
| 17:09.00 | dan_j | hmm. ok. |
| 17:09.20 | dan_j | thought I may be missing something obvious. weird that no one thought to display the useragent |
| 17:11.12 | mknooihuisen | Iâm dynamically generating speech via TTS and inserting it into the sounds directory. The message plays correctly (if a bit quietly) but I receive a warning: âWARNING[2638][C-0000000c]: mp3/interface.c:217 decodeMP3: Junk at the beginning of frame 49443304â and Iâm wondering if that will affect quality, and how to resolve it. Any thoughts? |
| 17:12.13 | igcewieling | mknooihuisen: That is the ID3 tag, asterisk doesn't understand it. you really should be saving .wav or something like that and stay away from mp3 |
| 17:14.00 | *** join/#asterisk andrelcm (~andre.mag@177.69.178.125) |
| 17:16.50 | mknooihuisen | igcewieling: I was hoping to avoid wav just due to their size, but I suppose I can manage it as long as I have sane logic for removing old files that are no longer needed. |
| 17:16.55 | mknooihuisen | Thanks |
| 17:18.59 | [TK]D-Fender | You're compromising your quality going through MP3 |
| 17:21.37 | mknooihuisen | True, but even the best quality is still going to be text-to-speech⦠at least, so went my logic. |
| 17:21.50 | *** part/#asterisk metanovii (~metanovii@176.49.170.21) |
| 17:21.59 | *** join/#asterisk metanovii (~metanovii@176.49.170.21) |
| 17:22.32 | metanovii | dudes, i try run asterisk after compile . and have this error - codec_opus_config.c: Opus: failed to register sorcery object type 'opus' . Anybody know what wrong? |
| 17:23.29 | skyroveRR | Which asterisk version is it, metanovii ? |
| 17:24.09 | metanovii | 13.13.1 |
| 17:24.40 | skyroveRR | Pastebin the build log. |
| 17:26.12 | metanovii | but build finished without troubles |
| 17:29.41 | rmudgett | dan_j: Looks like the AMI PJSIPShowEndpoint action includes the user agent as one of the headers in the ContactStatusDetail event. |
| 17:30.01 | *** join/#asterisk tzafrir_ (~tzafrir@37.26.147.138) |
| 17:30.57 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 17:32.59 | dan_j | Thanks |
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| 18:01.13 | mknooihuisen | Now Iâm getting an âUnable to open format wavâ but Iâm (almost) positive I enabled wav when I ran make menuselect |
| 18:01.59 | mknooihuisen | Do I need to rebuild? |
| 18:03.47 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-14.wall.uls.co.za) |
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| 18:04.45 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 18:08.44 | skyroveRR | [TK]D-Fender: hey, figured out the issue :) |
| 18:20.38 | igcewieling | mknooihuisen: no. |
| 18:20.47 | igcewieling | show us the line generating the issue |
| 18:22.37 | mknooihuisen | igcewieling: dialplan is: same => n,Playback(folder/test2) |
| 18:23.13 | mknooihuisen | Error is: [Feb 6 12:58:28] WARNING[2689][C-0000000d]: file.c:472 fn_wrapper: Unable to open format wav |
| 18:23.44 | igcewieling | you didn't so something stupid and try paring down your asterisk by noload=> in modules, did you? |
| 18:24.12 | mknooihuisen | Nope |
| 18:24.50 | mknooihuisen | But if there is a command to check the status of a module, Iâd run it |
| 18:25.12 | igcewieling | module show like wav |
| 18:25.14 | [TK]D-Fender | skyroveRR, And what was it? |
| 18:26.15 | skyroveRR | [TK]D-Fender: it's a known bug in the bluetooth/USB subsystem within the linux kernel; dmesg is flooded with these lines: Bluetooth: hci0 SCO packet for unknown connection handle 0 |
| 18:26.20 | mknooihuisen | igcewieling: format_wav and format_wav_gsm both running |
| 18:26.37 | [TK]D-Fender | skyroveRR, Glad you found it. I've never personally used BT before |
| 18:26.44 | [TK]D-Fender | on linux |
| 18:27.08 | skyroveRR | [TK]D-Fender: so basically everytime an SCO packet is sent to/from the phone, the kernel starts to scream. |
| 18:27.16 | igcewieling | what does "file /var/lib/asterisk/sounds/en/folder/test2.wav" tell you |
| 18:28.49 | hdon | hi all :) when i see "C-xxxxxxxx" where "x" is a hex digit -- does that identify a channel? |
| 18:29.25 | hdon | example: [2017-02-06 10:27:29] VERBOSE[3670][C-0000173b] from my log |
| 18:30.00 | [TK]D-Fender | skyroveRR, SCO making Linux Kernel scream? Nothing ironic about that at all..... |
| 18:32.21 | rmudgett | hdon: No. That is an asterisk callid for associating log messages with the channel that initiated a call. |
| 18:32.25 | *** join/#asterisk jjrh (~weechat12@2607:f0b0:8:8035:796f:592e:3a21:a6eb) |
| 18:33.21 | hdon | thanks rmudgett |
| 18:34.40 | hdon | when i look at the output of "sip show channelstats" the "Call ID" column contains an 11-digit hex string that appears random, not serial, unlike the "C-xxxxxxxx" 8-digit call IDs. is it possible to correlate the two? |
| 18:36.24 | rmudgett | No. Those are two different things. |
| 18:37.06 | rmudgett | The C-xxxxxxxx is an Asterisk created identifier useful for grepping the logs. |
| 18:37.45 | hdon | yes it does seem to be useful for that |
| 18:37.49 | hdon | thanks again rmudgett |
| 18:37.52 | rmudgett | The "sip show channelstats" call-id is a SIP identifier that is part of the SIP protocol |
| 18:38.10 | hdon | ahhh |
| 18:39.18 | mknooihuisen | igcewieling: Sorry, missed your message. It shows: /var/lib/asterisk/sounds/en/folder/test2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz |
| 18:40.49 | igcewieling | you need to convert it to something asterisk likes, not-auth-pstn.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 18:40.58 | rmudgett | hdon: I should say that the C-xxxxxxxxx was created specifically so you could grep the logs for messages associated with a call. |
| 18:41.25 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 18:41.30 | hdon | rmudgett, mmm. a worth feature, to be sure. |
| 18:41.34 | hdon | worthy* |
| 18:42.19 | rmudgett | You won't necessarily get all messages though because some messages don't know what call they are dealing with at the time. |
| 18:42.43 | igcewieling | rmudgett: that is a lot less useful because we can't get the CXXXX info from the dialplan |
| 18:43.37 | igcewieling | I'll need a to write to script to find the call in the Asterisk logs, get the C-XXXXX info for the call, then grepping for it |
| 18:44.18 | hdon | rmudgett, yikes... |
| 18:47.24 | mknooihuisen | igcewieling: is it just the sample rate and mono vs. stereo that it doesnât like? Because Iâm not seeing any other differences between what I showed you and running file on one of asteriskâs sample .wav files |
| 18:47.35 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
| 18:48.19 | igcewieling | both |
| 18:48.58 | igcewieling | phones don't support stereo and don't support better than 8Khz audio (until recently) so there is no reason for Asterisk to support them. |
| 18:49.37 | mknooihuisen | Sure, makes sense. Thatâs an easy fix too |
| 18:49.45 | *** join/#asterisk bayan (~dootdoot@unaffiliated/bayan) |
| 18:58.03 | mknooihuisen | igcewieling: Thanks. Got it working |
| 18:58.22 | mknooihuisen | Fixed the volume issue the MP3s were having, too |
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| 21:18.29 | mknooihuisen | HI all. As Iâm setting up my dialplan I am noticing that I hear popping sounds somewhat randomly either before or after using Playback. The error isnât very reproducible, as it only happens some of the time, and can happen on any audio file, including asteriskâs default sounds. Any ideas what is causing this? |
| 21:24.10 | mknooihuisen | Ah, well, never mind for now. Something came up and I canât stick around. Will probably revisit this later. Thanks for all the help getting set up so far! |
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| 23:11.02 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 23:11.07 | Kobaz | <PROTECTED> |
| 23:11.55 | Kobaz | this lady is on a polycom 331, she's got a 3.5mm headset... when she switches to headset everything her far-end says gets echoed back to the far end |
| 23:12.05 | Kobaz | so for instance, i call her, everything i say echos back to my own phone |
| 23:12.17 | Kobaz | so... bad headset you say? |
| 23:12.26 | igcewieling | that can be the speaker on the headphone bleeding into the mic. |
| 23:12.32 | Kobaz | new phone, polycom 335, new headset |
| 23:12.41 | Kobaz | same thing occurs |
| 23:13.26 | igcewieling | um, the user can crank the volume back to REALLY LOUD as soon as the phone is replaced. Does this exact make/model of headset work elsewhere? |
| 23:13.35 | igcewieling | Polycom can be picky about the headsets used. |
| 23:13.57 | Kobaz | yeah |
| 23:14.01 | WIMPy | Or a bad plug/cable. |
| 23:14.15 | Kobaz | it's a basic panasonic 3.5mm headset |
| 23:14.24 | Kobaz | like $9 on amazon |
| 23:14.32 | Kobaz | new headset, new phone, same problem |
| 23:14.43 | Kobaz | she's on the same edge vpn gateway though |
| 23:15.15 | Kobaz | i can hear my audio in her rtp |
| 23:15.25 | Kobaz | rtp from polycom -> server. i hear my voice back |
| 23:15.30 | Kobaz | and this is captured on the vpn box |
| 23:16.14 | [TK]D-Fender | Kobaz> so... bad headset you say? <- clearly |
| 23:16.25 | Kobaz | clearly |
| 23:16.34 | Kobaz | until you put in a different headset on |
| 23:16.53 | Kobaz | that was tested successfully on another phone |
| 23:17.09 | [TK]D-Fender | maybe the pther phone was weak |
| 23:17.17 | [TK]D-Fender | Because that'd only be ACOUSTIC echo |
| 23:17.17 | Kobaz | weak? |
| 23:17.27 | [TK]D-Fender | weak OUTPUT or oversensitive mic |
| 23:17.32 | Kobaz | ah |
| 23:17.42 | Kobaz | weak output maybe |
| 23:17.47 | Kobaz | i can barely hear her when she goes on headset |
| 23:17.55 | Kobaz | but i hear my own voice echoed back really loud |
| 23:18.15 | [TK]D-Fender | Or crossed wires |
| 23:18.44 | Kobaz | same crossed wires on a brand new 335? and a brand new headset? |
| 23:18.50 | Kobaz | and! |
| 23:18.53 | Kobaz | to make matters worse |
| 23:19.02 | Kobaz | i tested this, in my office with no issues prior to mailing this out |
| 23:19.30 | Kobaz | she's been having these echo issues with the 331/headset combo for a few weeks... trying to troubleshoot remotely blah blah blah |
| 23:19.48 | Kobaz | so i configed a new 335, took a headset out from a box, plugged it in, made some calls... worked fine |
| 23:20.03 | Kobaz | mail it down to florida and now it gets the same echo problem as the 331 had |
| 23:21.24 | [TK]D-Fender | What model are you getting them, and for what kind of usage? |
| 23:22.01 | Kobaz | https://www.amazon.com/Panasonic-KX-TCA60-Hands-Free-Headband-Cordless/dp/B00007M1TZ |
| 23:22.10 | Kobaz | it's your super basic run of the mill garden variety headset |
| 23:22.16 | Kobaz | this customer is cheap (yeah... i know) |
| 23:23.11 | Kobaz | so |
| 23:23.12 | Kobaz | wtf |
| 23:23.20 | Kobaz | works fine in new york, ship to florida, same issue |
| 23:23.26 | Kobaz | same issue as the existing phone |
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