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| 01:19.24 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); 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 |
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| 09:46.45 | Amnesia | question, is it correct that a client's status, is a setting that's only got effect locally ? |
| 09:46.56 | Amnesia | (local from the clients perspective) * |
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| 12:04.26 | jamesaxl_05 | hi |
| 12:05.28 | jamesaxl_05 | How can I configure asterisk to start phone timer only if client answer? |
| 12:06.34 | Samot | What do you mean? |
| 12:06.38 | Samot | What phone timer? |
| 12:06.51 | jamesaxl_05 | Samot: I meant duration |
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| 12:07.04 | Samot | What duration? |
| 12:07.06 | Samot | Be specific. |
| 12:07.44 | jamesaxl_05 | Samot: Ok, I want to call someone, and I want the duration start in my softphone only if client answer |
| 12:07.59 | Samot | Which duration? |
| 12:08.06 | Samot | What do you mean? |
| 12:08.09 | Samot | You mean in the CDR? |
| 12:08.13 | Samot | You mean a sip timer? |
| 12:08.16 | Samot | What duration? |
| 12:09.26 | jamesaxl_05 | Samot: in cdr we have duration and billsec. I want to see in softpone just billsec |
| 12:09.48 | Samot | Duration is how long the call was in asterisk... |
| 12:09.57 | Samot | billsec is the call after it was _answered_ |
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| 12:10.38 | jamesaxl_05 | yes exactly, and in the phone when I call to someoe here the timer start even if he/she does not answer |
| 12:10.43 | Samot | And if you think you can magically make your softphone's timer change, you're wrong. |
| 12:10.57 | Samot | That's now how those work. |
| 12:11.42 | Samot | It's also not even close to an Asterisk thing either. |
| 12:12.54 | jamesaxl_05 | but why those employees told me that the system before was like ..... Strange I have also same feeling, cause there is no relation between asterisk and phones |
| 12:14.12 | jamesaxl_05 | Samot: I am sorry, and thank you for help |
| 12:14.54 | Samot | So the users are saying that the old phone system would only show the call duration on a random softphone if the call was answered... |
| 12:15.03 | Samot | What was the old system? |
| 12:15.16 | jamesaxl_05 | Samot: Freepbx |
| 12:15.21 | Samot | No. |
| 12:15.34 | Samot | Are these calls they are dialing themselves? |
| 12:16.14 | jamesaxl_05 | Samot: I really do not know anything about old system. |
| 12:16.18 | Samot | Are these calls they are dialing themselves? |
| 12:16.45 | Samot | Or is this some automated outbound calling thing? |
| 12:16.53 | JohnWigley | Could this be that the old system worked properly in the sense that the call wasn't answered unless the remote party answered it, but the current system has a strange behaviour where all outbound calls are answered immediately and then a ringback tone is synthesised? I've seen that before on for eg calling card services. That would explain the behaviour jamesaxl_05 is seeing. |
| 12:17.00 | Samot | It's FreePBX |
| 12:17.26 | Samot | It's not a normal behavior |
| 12:17.31 | Samot | Because it's still _Asterisk_ |
| 12:17.32 | Samot | So again |
| 12:17.54 | Samot | Are these calls the users are dialing via the keypad or is this some automated outbound calling process? |
| 12:18.06 | Samot | Because that _does_ make a difference. |
| 12:19.23 | jamesaxl_05 | Samot: nop, workers enter the number |
| 12:19.32 | Samot | So they are dialing? |
| 12:19.37 | Samot | Then it never did that. |
| 12:19.38 | Samot | Ever. |
| 12:19.43 | jamesaxl_05 | Samot: yes |
| 12:19.46 | Samot | They are wrong. |
| 12:20.17 | Samot | Where they using the same softphone? |
| 12:20.40 | jamesaxl_05 | Samot: yes we do not change them |
| 12:20.49 | Samot | What softphone? |
| 12:21.03 | jamesaxl_05 | JohnWigley: I think originate does like that |
| 12:21.16 | jamesaxl_05 | Samot: phonerlite and linphone |
| 12:21.25 | Samot | That's not what FreePBX does. |
| 12:21.31 | Samot | Because FreePBX is just Asterisk. |
| 12:21.56 | Samot | With a GUI and a few extra things. |
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| 12:22.30 | JohnWigley | jamesaxl_05: How are your calls actually connecting to the PSTN, through an ITSP or a local gateway? |
| 12:22.43 | jamesaxl_05 | Samot: I heve to ask other workers, and I must have more info |
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| 12:23.29 | jamesaxl_05 | JohnWigley: ITSP |
| 12:24.16 | JohnWigley | So then I would try taking the ITSP trunk details, putting that into a softphone and seeing if you get the same behaviour, if so then it's the ITSP answering the call immediately and synthesizing ringback, and Asterisk is just passing it through. |
| 12:24.30 | Samot | What? |
| 12:24.31 | Samot | No. |
| 12:24.37 | Samot | That's not how that works at all. |
| 12:24.45 | Samot | The softphone only knows of ASTERISK |
| 12:24.57 | Samot | Asterisk is what is _answering_ the call from the softphone. |
| 12:24.59 | Samot | Not the ITSP |
| 12:25.09 | Samot | That call could never leave Asterisk... |
| 12:25.18 | JohnWigley | Sorry are these for internal calls? |
| 12:25.21 | Samot | It only does because there's dialplan that says "Dial() |
| 12:25.34 | JohnWigley | I misunderstood and thought we were talking about external calls. |
| 12:25.41 | Samot | It doesn't matter.. |
| 12:26.00 | jamesaxl_05 | I am going to ask workers for getting more info. |
| 12:26.24 | JohnWigley | Samot: Of course it does, if user dials on softphone, and asterisk dials the ITSP, if the ITSP answers immediately and synthesizes ringback, then the softphone will see the call answered immediately and start the call duration timer. |
| 12:26.51 | JohnWigley | It's not by any means what is actually happening, but it would exactly explain the behaviour he's seeing. |
| 12:27.10 | JohnWigley | Trying an internal call to see if this happens, would also eliminate this as a possibility however unlikely it is. |
| 12:28.43 | Samot | Yes, it's the dialplan. |
| 12:28.59 | Samot | It's not the ITSP, it's how the Dial() is being handled. |
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| 14:09.52 | kai[El] | tr/list |
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| 14:20.29 | mneth1 | I switched my ast13 deployment from chan_sip to pjsip and CID matching in my dialplans doesn't work now. Even if I try to use _! to catch all CIDs, if I have any CID match pattern then it doesn't match anything. Has anyone run into this? |
| 14:21.47 | [TK]D-Fender | Show us the config & call |
| 14:23.42 | mneth1 | One moment, let me sanitize them |
| 14:28.29 | [TK]D-Fender | Don't |
| 14:28.51 | [TK]D-Fender | If you literally screw with the evidence then we're likely going to end up wasting our time |
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| 14:31.12 | mneth1 | hm yeah, I just have to make sure I don't have company or customer info in the data. I'll try to make sure it's enough info. Here are the pattern matching lines, hopefully pastebin is ok: https://pastebin.com/pqTiFTXh |
| 14:32.28 | mneth1 | Basically, if I try to match on CID then it doesn't work with PJSIP. chan_sip worked just fine. Example call info that isn't working on pjsip but works on chan_sip - Exten: +18004444444, CID: +13160000000 |
| 14:33.07 | [TK]D-Fender | Show actual config and an actual call |
| 14:34.34 | mneth1 | If I don't add in the pattern match that leaves out the CID part, incoming calls just get this output: res_pjsip_session.c:2132 new_invite: Call from 'provider' (UDP:x.x.x.x) to extension '+18004444444' rejected because extension not found in context 'incoming'. |
| 14:34.57 | mneth1 | hm let me see what else I can show, after those pattern match lines it just dumps it into the routing logic |
| 14:38.49 | *** join/#asterisk NonSecwitter (~NonSecwit@unaffiliated/nonsecwitter) |
| 14:43.22 | mneth1 | Here's the output of 'dialplan show' for the incoming context https://pastebin.com/jJ7fMAtg |
| 14:43.53 | mneth1 | then calling +18004444444 from +13160000000 results in the call not connecting and this is the only output in asterisk: res_pjsip_session.c:2132 new_invite: Call from 'provider' (UDP:x.x.x.x) to extension '+18004444444' rejected because extension not found in context 'incoming'. |
| 14:44.31 | mneth1 | Seems to be the same issue this guy was experiencing but I haven't found any open issues on the tracker for it: http://asteriskfaqs.org/2017/08/02/asterisk-users/callerid-matching-failure-possibly-bug.html |
| 14:52.46 | [TK]D-Fender | taht isn't an actual bug report and we aren't looking at anything real |
| 14:52.50 | [TK]D-Fender | this is wasting our time |
| 14:54.20 | mneth1 | What other info can I give you to help? |
| 14:54.31 | [TK]D-Fender | [TK]D-Fender> Show actual config and an actual call |
| 14:56.03 | mneth1 | Was that `dialplan show` output I linked above not enough for the config? Maybe i'm unsure what you're asking for. |
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| 14:56.35 | *** mode/#asterisk [+o kharwell] by ChanServ |
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| 15:02.20 | [TK]D-Fender | WHERE'S THE FULL CALL DEBUG? |
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| 15:05.15 | *** join/#asterisk bounceman (~bounceman@185.32.9.250) |
| 15:05.49 | bounceman | Hello, can anyone tell me the difference between simple-bridge and softmix bridging technologies? |
| 15:07.04 | mneth1 | Here's the debug I believe you're looking for: https://pastebin.com/eWE4U0Pz |
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| 15:10.45 | NonSecwitter | in a conference call, do all participants share a call-id, with dialogs uniquely identified by tags? |
| 15:14.38 | [TK]D-Fender | You're clearly editing out details |
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| 15:14.43 | [TK]D-Fender | I'm done |
| 15:14.51 | [TK]D-Fender | moves on to more productive matters |
| 15:15.14 | jamesaxl_05 | [TK]D-Fender: why? |
| 15:15.24 | mneth1 | I just removed IPs. |
| 15:16.00 | jamesaxl_05 | [TK]D-Fender: you should be patient |
| 15:17.27 | [TK]D-Fender | jamesaxl_05, I asked for configs and the call AN HOUR AGO. |
| 15:17.40 | [TK]D-Fender | Told him NO FILTERING |
| 15:17.42 | [TK]D-Fender | I'm done |
| 15:18.03 | jamesaxl_05 | feels sad |
| 15:20.18 | bounceman | [TK]D-Fender: you can ask me anything, I just want to know about the two different bridge technologies |
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| 15:31.57 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:33.34 | jamesaxl_05 | Samot: you know, whithout function Answer the counter of the softphone starts after client answers. |
| 15:33.54 | jamesaxl_05 | but I can not hear cient, ad client also can not hear me |
| 15:33.56 | Samot | Right. |
| 15:34.12 | Samot | I said it was in the dialplan and how the dial was handled. |
| 15:34.52 | jamesaxl_05 | Samot: is it possible to answer channel after Dial? |
| 15:35.37 | Samot | If the otherside answers, it is answered. |
| 15:35.55 | Samot | If they don't then you can handle the call based on the DIALSTATUS |
| 15:37.09 | jamesaxl_05 | Samot: I test it with my phone, but i do not hear my self, but if I put Answer, I can hear |
| 15:37.49 | Samot | Then there is something up with the dialplan or how the system is handling NAT or something. |
| 15:37.58 | Samot | Nothing that I can just say "this is it" to. |
| 15:39.03 | jamesaxl_05 | Samot: I am not using nat, my server directy connected to Internet. Strange I am going to check |
| 15:39.13 | jamesaxl_05 | internal calls works well |
| 15:39.42 | Samot | This is a configuration issue. |
| 15:39.54 | Samot | FreePBX doesn't Answer() for things like this. |
| 15:40.02 | Samot | That's why it happens like it did in FreePBX. |
| 15:40.13 | Samot | It was how the dialplan and the configs were generated. |
| 15:42.08 | jamesaxl_05 | time to check |
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| 17:41.10 | jamesaxl | Home sweet Home :) |
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| 19:05.48 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); 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 |
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| 19:26.47 | mXr | good evening, anyone here who might be able to shed some light on how qualify/auth is supposed to work with pjsip for me? (asterisk 15.2.0) |
| 19:29.29 | mXr | im using sorcery/realtime config in db, i think i understand most of what im setting there, but .. when i try to set up one asterisk as REGISTER-ing "client" and another as "server", i dont understand how qualify should look like |
| 19:30.08 | mXr | with a normal softphone client, registering works, and so does the qualify on that client; it sends OPTIONS out, it gets OK back |
| 19:30.49 | mXr | when i use asterisk as the "client" tho, it sends out 401 unauthorized, the other side then complains about not being able to find matching auth for realm "asterisk", and both sides are unhappy |
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| 19:55.26 | mXr | also, what does authenticate_qualify do? i searched thru the source and all i can find is option handling code. as if the option existed, but actually didnt do anything at all |
| 19:59.34 | *** topic/#asterisk by gtjoseph -> #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 |
| 19:59.59 | mXr | ok it seems it did do something in the past, but no longer so https://gerrit.asterisk.org/#/c/2373/9/res/res_pjsip/pjsip_options.c |
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| 20:38.31 | elcontrastador | Do polycom vvx cameras work with Asterisk? |
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| 20:46.21 | [TK]D-Fender | elcontrastador, if they talk SIP then yes. |
| 20:46.51 | elcontrastador | cool...thank you. Yes, will use lastest Asterisk and pjsip |
| 20:47.15 | Samot | VVX cameras attach to VVX phones. |
| 20:47.29 | Samot | The camera's themselves do not talk anything on their own. |
| 20:48.35 | Samot | You can have 3 SIP accounts on your VVX phone that all support video. The camera will not care which account you use. |
| 20:48.58 | elcontrastador | got it...just didn't want to waste money with any known issues |
| 20:49.04 | elcontrastador | thanks guys |
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| 20:50.33 | Samot | Well just remember, VVX50x VVX60x are the phones those cameras are for.. |
| 20:50.39 | Samot | The VVX1500 has a built in camera |
| 20:51.16 | Samot | The lower models don't support video. Though I haven't really checked if the 401/411's do since they got a hardware boost. |
| 20:53.56 | elcontrastador | it's only for 601 models |
| 20:54.12 | elcontrastador | <my potential deployment> |
| 21:29.28 | talntid | When you guys sell phone systems, where do you typically source your phones from? do you have partnerships with people like VoipSupply? |
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| 22:33.47 | *** 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 |
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| 22:43.15 | drmessano | I hear VoIP Supply has great prices on Sangoma phones |
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| 22:47.17 | Samot | Do tell. |
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