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| 00:58.45 | dl12 | hi guys i need help with a patch, never done this before |
| 00:59.10 | dl12 | the confbridge app need to be patched to make the join/leave fonction work |
| 01:02.36 | [TK]D-Fender | confirm what you're running, show us the configs, and show us the call which fails it |
| 01:05.24 | dl12 | asterisk 13.15.0 |
| 01:05.43 | dl12 | https://pastebin.com/6v1KM1KW |
| 01:05.54 | dl12 | the join/leave option not working |
| 01:07.44 | dl12 | this confirm https://gerrit.asterisk.org/#/c/5683/ |
| 01:08.21 | dl12 | the name file is deleted before the playback |
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| 01:12.46 | [TK]D-Fender | LTS: 13.19.1 (2018/02/13) |
| 01:12.55 | [TK]D-Fender | You're quite behind |
| 01:13.01 | [TK]D-Fender | upgrade and reproduce first |
| 01:14.00 | dl12 | can i upgrade without loosing my configuration? i never upgraded before |
| 01:22.17 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:22.17 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.1 (2018/02/13), Standard: 15.2.1 (2018/02/13); 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 |
| 01:24.31 | [TK]D-Fender | maybe something in the upgrades helps your situation. |
| 01:45.34 | *** join/#asterisk Mr_Pleb_Mgoo (af2d55ca@gateway/web/freenode/ip.175.45.85.202) |
| 01:50.27 | Mr_Pleb_Mgoo | anyone know what might be happening here... having some issues with ${DIALSTATUS} returning blank/empty values. calls are generated by callfiles placed in the spool folder. short pastebin showing some details https://pastebin.com/q1KS4UJA |
| 01:51.05 | [TK]D-Fender | Thre is no dialstatus |
| 01:51.10 | [TK]D-Fender | YOU didn't call Dial() |
| 01:51.37 | [TK]D-Fender | and the only reason you're IN the dialplan is because that SIP device you called actually answered |
| 01:51.45 | [TK]D-Fender | So even checking it is completely pointless |
| 01:51.57 | [TK]D-Fender | You're IN an answered channel at that point |
| 01:52.12 | [TK]D-Fender | You don't seem to have thought this through :) |
| 01:52.15 | Mr_Pleb_Mgoo | so dial.c saying a called was answered doesn't update dialstatus? |
| 01:52.35 | [TK]D-Fender | Dialstatus is also only at the END of a call |
| 01:52.38 | [TK]D-Fender | you're IN one |
| 01:52.50 | file | it's also a dialplan variable from the Dial() application |
| 01:52.55 | [TK]D-Fender | And again ... just because * dialed... Dialstatus is a DIALPLAN APP variable |
| 01:53.01 | [TK]D-Fender | you're not calling the app |
| 01:53.17 | [TK]D-Fender | the function the app uses may have done the dirty work for your originate but it doesn't apply here |
| 01:53.40 | Mr_Pleb_Mgoo | [TK]D-Fender: this is a system that has just sat there and worked for a very long time. im going to guess something has minced this server/dialplan |
| 01:53.45 | [TK]D-Fender | And again.. you're IN the call at that point. |
| 01:53.47 | Mr_Pleb_Mgoo | sigh |
| 01:54.09 | [TK]D-Fender | So that idea would never work or serve any purpose the way you're doing it |
| 01:54.56 | [TK]D-Fender | The logic is just wrong |
| 01:56.29 | Mr_Pleb_Mgoo | looks like somebody tried to add that in but never applied it, and when the server crashed from a DC outage it's obviously loaded it... i had a feeling there was something off about it |
| 01:56.39 | Mr_Pleb_Mgoo | applied = reloaded config* |
| 01:57.00 | Mr_Pleb_Mgoo | good times :) |
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| 02:12.58 | kfife | sip show channelstats: What's the best way to measure a burst of packet loss? In other words a call could be unacceptable at times while the average loss staying within an otherwise acceptable range. |
| 02:13.54 | kfife | I can poll these stats over AMI, and follow state, trying to see if loss is climbing at a meaningful rate over the last poll interval. Is there a better way? |
| 02:15.20 | kfife | I don't want our salespeople to make VoIP calls over their shitty home Wi-Fi, sounding like crap to the customers. |
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| 02:27.58 | [TK]D-Fender | That's the only natural way I can think of. The next might be an actually loaded module of your own code to quey more directly from the running process |
| 02:29.49 | kfife | [TK]D-Fender: sounds right to me. |
| 02:35.04 | kfife | [TK]D-Fender : thanks. If I had some time, I'd do that. This channelstats report could have configurable metric for MOS, or subjective quality, derived from patterns in the packet loss (e.g bursts of loss) with thressholds for different coders. i.e ILBC with loss concealment, vs simple PCM vs g.722 with loss amplificaition) |
| 02:35.29 | kfife | For us, we may just track stat in Node, raise an alarm if a rep starts to suck ass. |
| 02:36.15 | kfife | MOS=Mean Opinion Score |
| 02:36.22 | kfife | SAF=Suck Ass Factor |
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| 05:25.47 | jamesaxl | Hello |
| 05:25.49 | jamesaxl | this is my macro that I use for recording incoming calls https://bpaste.net/show/7fc9256d4607. My problems is : when I have transfer from operator to an other queue, the monitor inherit auto and the calls is recorded in the folder of the operator that did the transfer, so I can have the path of video file in data base but I can have any audio file in the last operator that took the call. Could you suggest |
| 05:25.51 | jamesaxl | for me a solution please |
| 05:25.53 | zenmaster | Hi. |
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| 06:18.21 | klow | Hey.. I'm trying to use the asterisk rest interface for the first time , just using curl. Trying this results in a JSON response saying "Resource not found" : curl -u asterisk:asterisk -X GET http://localhost:8088/ari/asterisk/info |
| 06:18.50 | klow | am I missing something? If I try other urls they are different.. 404 error , and if I put in different credentials, it doesnt even get this far |
| 06:19.03 | klow | it seems my http server is on, ari is activated (I think) , and my authentication with curl is working |
| 06:19.18 | klow | but still am getting resource not found trying to access several different API urls |
| 06:20.22 | klow | seeing this in my log: res_ari.c: Handler not found |
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| 13:41.14 | dl12 | hello fender, I upgrade to 13.19 this morning |
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| 13:41.36 | dl12 | im not sure i can ask here but since the upgrade i lsot some freepbx menu tab, they are grayed out saying asterisk is not running |
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| 14:56.15 | DivideBy0 | klow: Try turning debug on and seeing what it tells you. Did you set a path for ari in the config file? It might be wrong |
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| 15:31.08 | dl12 | is it possible to do this : I access my mailbox and instead of asking to dial 1 to play message it will play them directly? |
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| 15:34.20 | [TK]D-Fender | no |
| 15:35.55 | dl12 | ok |
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| 15:52.16 | comrad | hi there |
| 15:52.23 | comrad | i have a question to pjsip |
| 15:52.43 | Samot | Ask. |
| 15:53.19 | comrad | why do the id in ps_aors, ps_auths and ps_endpoints be equal to my username in ps_auths? |
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| 15:53.44 | comrad | i've just trying it, and it only works if the username is the same as the id in these three tables |
| 15:53.58 | comrad | "it works" means my telefone can register with the asterisk server |
| 15:54.38 | Samot | Right. |
| 15:54.42 | Samot | It's relational. |
| 15:54.58 | Samot | auth <-- where the auth for the user |
| 15:55.01 | comrad | so my credentials have to match my sip-id? |
| 15:55.06 | Samot | ? |
| 15:55.10 | Samot | No, you set them. |
| 15:55.32 | comrad | ive put them in ps_auths in username and password |
| 15:55.46 | comrad | let's say "comrad" as username |
| 15:56.01 | comrad | now my id in ps_aors, ps_auths and ps_endpoint has to be "comrad" as well |
| 15:56.09 | comrad | otherwise i cannot register with the server |
| 15:56.27 | comrad | i fail to understand why this is coupled |
| 15:56.47 | Samot | 10:54:42 AM S<Samot> It's relational. |
| 15:56.54 | Samot | The Endpoint is the actual device. |
| 15:57.01 | Samot | Which has to be bound to a user |
| 15:57.16 | Samot | Auth <-- Is the user details. |
| 15:57.26 | comrad | yes |
| 15:57.34 | comrad | and it is linked in ps_endpoints.auths |
| 15:57.37 | Samot | AORs <-- Tell's it how many contacts that user can have. |
| 15:58.00 | Samot | Since PJSIP will allow for multiple contacts to register as the same user. |
| 15:58.11 | Samot | So it's all relational. |
| 15:58.20 | comrad | and still the ids have to match my username |
| 15:58.26 | Samot | Yes. |
| 15:58.36 | comrad | why enter the username at all? |
| 15:58.48 | comrad | or why have ids at all if they are not used? |
| 15:59.59 | Samot | I don't know what you are asking. |
| 16:02.58 | comrad | ps_aors -> id (lets say id is "helloworld") |
| 16:02.58 | [TK]D-Fender | tha AOR does not have to have the sme name as the endpoint |
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| 16:02.58 | [TK]D-Fender | endpoint points to whatever aor's you tell it to |
| 16:02.58 | comrad | ps_auths -> id ("helloworld"), username ("myphone") |
| 16:02.58 | [TK]D-Fender | same for other structures |
| 16:02.58 | comrad | ps_endpoints -> id ("helloworld"), aors ("helloworld"), auths ("helloworld") |
| 16:02.58 | comrad | this setup wont work untill i change username to "helloworld" as well |
| 16:02.58 | Samot | ARe you talking about the [section] names? |
| 16:02.58 | comrad | asterisk realtime |
| 16:02.58 | Samot | Right |
| 16:02.58 | Samot | So those are the "sections" |
| 16:02.58 | comrad | yeah its pretty much the same |
| 16:02.58 | Samot | It's a multi-piece config. |
| 16:02.58 | comrad | i know that |
| 16:02.58 | Samot | SO |
| 16:02.59 | Samot | REALTIONAL |
| 16:03.07 | Samot | How do the multi-parts know about each other? |
| 16:03.12 | Samot | They have the same NAME |
| 16:03.17 | Samot | With different DATA |
| 16:03.20 | comrad | they should reference via id |
| 16:03.26 | Samot | What ID? |
| 16:03.31 | comrad | well, the field ID |
| 16:03.41 | comrad | ps_aors.id |
| 16:03.43 | comrad | ps_auths.id |
| 16:03.44 | Samot | OK.. |
| 16:03.47 | comrad | ps_endpoints.id |
| 16:03.50 | Samot | That's a RealTime thing. |
| 16:03.52 | comrad | they should be relational |
| 16:03.59 | Samot | RealTime is secondary |
| 16:04.07 | Samot | It's not the main deployment method of Asterisk. |
| 16:04.30 | Samot | So RealTime is the way it is to support how PJSIP works statically |
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| 17:06.17 | nny | This is odd, thoughts appreciated. I have two servers across an IPSEC tunnel that one stopped being able to register to the other at 10 PM last night. Nothing supposedly changed topology wise, B can register to A but A can't to B. I see 401 unauths at B regarding A both in packet dumps and sip debug, but cannot ascertain how this just randomly started late last night |
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| 17:23.29 | jamesaxl | this is my macro that I use for recording incoming calls https://bpaste.net/show/7fc9256d4607. My problems is : when I have transfer from operator to an other queue, the monitor inherit auto and the calls is recorded in the folder of the operator that did the transfer, so I can have the path of video file in data base but I can have any audio file in the last operator that took the call. Could you suggest |
| 17:23.30 | jamesaxl | for me a solution please |
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| 17:56.49 | zaf | jamesaxl, how about you don't set the recording up for the operator, and wait to do that on the non-operator extensions? |
| 17:58.37 | jamesaxl | zaf: could you explain more please |
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| 19:20.12 | nny | whats the option to perform an action on connected with Dial |
| 19:22.37 | nny | G? |
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| 19:25.39 | UncleKiwi | hi - chan_iax2.c:5082 handle_call_token: Too much delay in IAX2 calltoken timestamp from address x.x.x.x |
| 19:26.06 | UncleKiwi | i don't know why my system isbeing haunted by this error |
| 19:39.41 | tzafrir | The call token is right at the start of the call, right? |
| 19:39.57 | tzafrir | Does this mean there was nothing from the other side? |
| 19:43.08 | UncleKiwi | it has a cisco ASA that is maintained by a 3rd party IT crowd - Maybe they have down some config changes that have affected this... has been functioning for the last 230 days |
| 19:44.46 | UncleKiwi | asterisk is behind a NAT and is registering to a IAX2 VoIP provider |
| 19:45.02 | UncleKiwi | as mentioned it has been perfect for the last 230 days |
| 19:45.20 | Samot | Have you asked them? |
| 19:45.27 | UncleKiwi | yeah |
| 19:46.12 | UncleKiwi | Samot you mean the IT crowd ? |
| 19:46.43 | Samot | Yes. |
| 19:46.51 | Samot | Did you ask if they made changes? |
| 19:47.30 | UncleKiwi | very hard to get in communication with them - that is what i will attempt to do now |
| 19:48.07 | jamesaxl | Samot: how do you handle transferred calls and MixMonitor ? |
| 19:49.25 | Samot | jamesaxl: I'll be honest with you, in cases of when a client needs call center style functionality, I use FreePBX because all the work has been done. |
| 19:49.49 | Samot | jamesaxl: As you stated, the previous system was FreePBX and now you are trying to mimic it. |
| 19:54.16 | jamesaxl | Samot: not 100%, the way that FreePBX use for recording file I know it very well, I tried to do something better for me (not better than FreePBX) And I had issue when someone transfer call to queue. |
| 19:55.54 | jamesaxl | for example Client call 'A' and after 'A' transfers the call to 'B' so the recording file is 'incoming_from_client_to_a' but I need also 'incoming_from_client_to_b' |
| 19:57.20 | jamesaxl | I should move my brain 100% to find a solution. |
| 20:00.24 | Samot | Doesn't the recording continue? |
| 20:00.50 | Samot | If A transfer the call to B and the recording is already active, doesn't it still record the call with B? |
| 20:01.31 | jamesaxl | Samot: It continue, and I can get in the folder of operator 'A' , and I would like to 'cp' the file also in the folder of operator 'B' |
| 20:01.56 | Samot | OK, that has nothing to do with the actual recording process... |
| 20:02.04 | Samot | It's a _single_ file. |
| 20:02.12 | Samot | You named it when you started the recording. |
| 20:03.18 | jamesaxl | Samot: can I copy it to an other folder when transfer is done? |
| 20:03.19 | Samot | The recording should be associated in the CDR |
| 20:03.25 | Samot | It's a file. |
| 20:03.32 | Samot | You can do what you want like any other file. |
| 20:03.39 | Samot | That's post recording stuff. |
| 20:03.47 | Samot | Like executing a post recording script |
| 20:04.14 | Samot | But the CDR only associates one file. |
| 20:04.53 | jamesaxl | Samot: of course it is associated in the CDR. thank you for you advice a lot |
| 20:05.14 | Samot | What i mean is... |
| 20:05.27 | Samot | Either A or B's recording can be linked to the CDR |
| 20:05.47 | Samot | There's only one field option, unless you create others. |
| 20:06.13 | jamesaxl | Samot: did you see my Macro that I use for recording files? |
| 20:06.23 | Samot | No. |
| 20:06.32 | jamesaxl | a moment please |
| 20:06.36 | nny | trying to run a gosub on transfer or blind transfer, I assume I either have to use the functions in asterisk vs the SIP client/phone or change the context for when transfers happen vs regular calls |
| 20:07.32 | Samot | nny: What do you mean? |
| 20:07.43 | jamesaxl | Samot: https://bpaste.net/show/dd94cf39be5d |
| 20:07.53 | Samot | nny: Attended Transfers are based on the device's context. |
| 20:08.07 | Samot | If it's in that context, they can transfer to it because it's a call. |
| 20:08.43 | Samot | blind Transfer I believe also defaults to that but you can specific a TRANSFER_CONTEXT on the channel(s) for what context blind transfers use. |
| 20:10.01 | *** join/#asterisk nny_ (ad5db800@gateway/web/freenode/ip.173.93.184.0) |
| 20:11.24 | Samot | You run that macro every time? |
| 20:11.35 | Samot | You create the same directory all the time? |
| 20:11.53 | Samot | Then move, not copy, the original recording? |
| 20:11.56 | jamesaxl | Samot: yes I do |
| 20:13.37 | jamesaxl | Samot: it is not recommended? |
| 20:17.08 | Samot | Well |
| 20:17.24 | Samot | If you want it to be copied into multiple directories.. |
| 20:17.27 | Samot | No. |
| 20:17.36 | *** join/#asterisk pruonckk (~pruonckk@135-143-11-177.raimax.com.br) |
| 20:18.29 | jamesaxl | Samot: understood. |
| 20:27.10 | kfife | I have a subroutine that I frequently call from the dialplan to dip the LIDB for CNAM. If I want to utilize that subroutine from AMI, do I have to 'Originate' a new "call" to its context and start prioirty, or is there a better way, for example turn the routine into a macro, and call the macro directly, or call the routine directly somehow, etc. I think I'm missing a concept here. |
| 20:28.33 | nny_ | Samot: sorry got disconnected, trying to curl a url on transfers wether they be blind or attended (right now the sip client executes the transfer) |
| 20:29.38 | jamesaxl | kfife: could you show us the content of subroutine? |
| 20:33.49 | kfife | https://pastebin.com/uBW7bHjh Complete with doc. Use it if you want. Saves us hundreds per year over |
| 20:34.02 | igcewieling | kfife: you have to fake a call. It might be easier to port your subroutine to php or perl or some other language. |
| 20:36.39 | igcewieling | You don' |
| 20:36.48 | igcewieling | You don't skip toll free numbers? |
| 20:37.06 | nny_ | While I am at it I am trying to also send a packet to the originating channel for a call with additional SIP HEADER info, so SIPADDHEADER won't work |
| 20:42.46 | kfife | igcewieling: and use agi instead of a gosub within the dialplan? |
| 20:43.10 | kfife | or you mean just in general, don't use the code as it sits in Asterisk. |
| 20:43.18 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 20:43.57 | kfife | igcewieling: don't skip tf kinda because who cares |
| 20:44.32 | kfife | I can make a REST call from elsewhere to get LIDB, but I already have a CACHE in AstDB |
| 20:45.12 | kfife | I'll try the fake call idea. |
| 20:45.14 | kfife | Thanks |
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| 21:03.20 | kfife | DUH... |
| 21:04.04 | kfife | OR, ALTERNATIVELY, I could just use the database lookup routine that's already in AMI! :-) |
| 21:04.06 | kfife | Sheesh. |
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| 21:12.22 | *** mode/#asterisk [+o bford_] by ChanServ |
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| 21:43.19 | dawn | Hello, I need some help with my configuration. I have to asterisk boxes connected through openvpn using sip register. Both phones are able to place a call, but on one phone the user hears their own voice on their phone and the other user cannot. However the first user can hear the call. I have disabled SIP ALG on the router with no advantage. I have nat=yes, externip and localnet setup. If anybody could point me in the right dir |
| 21:43.23 | dawn | thank you advance |
| 21:54.25 | *** topic/#asterisk by kharwell -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.2 (2018/02/21), Standard: 15.2.2 (2018/02/21); 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 |
| 22:09.53 | kfife | dawn: you probably have a firewall rule blocking your media |
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| 22:30.00 | *** join/#asterisk rwb (~Thunderbi@65.183.151.121) |
| 22:47.09 | Primer | Or not setup the RTP port range and NAT bounce |
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| 23:01.32 | dawn | Thank you for your suggestion, the problem was with canreinvite needed setting to no ! |
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