| 11:50.59 | *** join/#asterisk infobot (ibot@rikers.org) |
| 11:50.59 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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 |
| 12:05.15 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 12:33.52 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
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| 12:49.15 | rfr__ | When migrating from an old * zaptel configuration to a new * dahdi configuration on a new machine do I need to run dahdi_genconf if I have already copied zapata.conf to chan_dahdi.conf? |
| 12:51.50 | [TK]D-Fender | you never need to run that if your configuration is in right for your card |
| 13:02.45 | *** join/#asterisk comrad (~comrad@holarse/core/comrad) |
| 13:02.49 | comrad | hello |
| 13:05.13 | comrad | does the function SIPAddHeader(My-Header: X) do the same as PJSIP_HEADER(add,My-Header)=X? |
| 13:09.16 | *** join/#asterisk AsteriskRoss (259d3426@gateway/web/freenode/ip.37.157.52.38) |
| 13:10.06 | [TK]D-Fender | for its module, should.... |
| 13:10.28 | rfr__ | [TK]D-Fender Ok thank you, that answer gets me one half-step closer. |
| 13:10.59 | [TK]D-Fender | What are you actually trying to do? |
| 13:11.48 | AsteriskRoss | I have a dialplan that sets MicMonitor in a precall handeler so that call recordings remain even if the call is transferred. However DYNAMIC_FEATURES does not run PauseMonitor on the outbound channel, even if defined in the precall handeler. What would be the best way to implement this? |
| 13:12.36 | rfr__ | I have a very old Asterisk 1.2 configration that runs as a POTS PBX that I am trying to upgrade. |
| 13:13.23 | rfr__ | It's a bit hard because I can only hook it up to the channel bank (Adit 600) on weekends to test. |
| 13:14.01 | rfr__ | THis last weekend it did not go so well. |
| 13:15.38 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 13:15.51 | rfr__ | The dahdi_cfg man page says that dahdi_cfg is ussually run by the dahdi init scripts. Is there a way to confirm that that's the case? |
| 13:16.55 | [TK]D-Fender | check your init script folder and whatever you're using to start them |
| 13:17.22 | [TK]D-Fender | if you change runlevels or whatever other trigger method you should see it fire. |
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| 13:37.03 | rfr__ | Do you mean check inside /etc/init.d/dahdi? |
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| 13:38.16 | [TK]D-Fender | If that's what it's called on your distro |
| 13:38.36 | [TK]D-Fender | It's been a while since I've had to think about exactly whre those are and are named by distro... |
| 13:41.06 | rfr__ | Ok, thanks. |
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| 14:18.52 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 15:18.58 | stefan27 | what's the telephony term for "no-answer-routing" ? |
| 15:19.38 | stefan27 | a general word for the kind of call-routing PBXes typically do when a call attempt to an extension fails (due to busy or timeout etc) |
| 15:19.41 | [TK]D-Fender | Telco's have used "no-answer-transfer" here |
| 15:20.06 | [TK]D-Fender | that's about as generic a name as it deserves |
| 15:20.09 | stefan27 | Hmm, so maybe there isn't a better phrase |
| 15:20.37 | stefan27 | at least no-answer-routing is somewhat descriptive |
| 15:20.42 | [TK]D-Fender | Says what it is... there really isn't much to name |
| 15:22.00 | igcewieling | the USA term is "call forward no answer" (CFNA), but forwarding is a more common term if you are talking abuot calls going to voicemail. |
| 15:23.36 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.w1.tele.crt1.net) |
| 15:36.43 | Samot | However, BUSY is not a NO ANSWER |
| 15:36.51 | Samot | So those are two different responses. |
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| 15:37.58 | Samot | So it would be "Call Forward Busy" and "Call Forward No Answer" |
| 15:40.11 | Samot | Plus there are two different types of "timeout". 1) Timeout with no 1XX reply i.e. the device didn't respond. 2) Timeout after 1XX reply i.e. it rang for X seconds and nothing picked up or no other response was given. |
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| 17:38.24 | jcpeters01 | Hello, would someone be willing to help me with a TO header issue I am having? It is passing extension@sip-provider instead of extension@mydomain |
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| 17:49.10 | Samot | If this is an outgoing request (outbound call to PSTN) then it should have number@sip-provider, that's where it is going to |
| 17:50.49 | jcpeters01 | My provider is telling me to change it to extension@mydomain |
| 17:51.25 | jcpeters01 | This is a Broadsoft Trunk trying to communicate to hosted seats on Broadsoft |
| 17:52.06 | Samot | The it should be the from domain not the to domain |
| 17:52.25 | Samot | Unless "mydomain" is pointing to the broadsoft server. |
| 17:52.43 | jcpeters01 | mydomain is not, through dns, pointing to the broadsoft server |
| 17:56.11 | jcpeters01 | It looks like I have to come from mydomain and go to mydomain for the 4 digit dial to work over the trunk |
| 17:56.29 | jcpeters01 | Otherwise, it says no such user here |
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| 17:59.15 | Samot | You can set the from domain in the peer settings but it will use what is in the host setting for the to domain |
| 17:59.45 | Samot | You could set the host to your domain and use the outbound proxy setting to send the requests to the sip provider domain |
| 18:00.31 | jcpeters01 | Yep, that is how I have it set. |
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| 18:02.43 | jcpeters01 | unless, would type=friend negate that? |
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| 18:59.41 | Worldexe | just found one of my peers set up like 'fromdomain=mydomain; host=mydomain' with 'mydomain' being put also to /etc/hosts |
| 19:00.21 | Worldexe | looks like just your case, jcpeters01 |
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| 19:18.35 | sibiria | anyone got any clues as to why CDR(billsec) and CDR(duration) are always the same in asterisk 13? |
| 19:18.52 | sibiria | even if you let it ring for several seconds before answering |
| 19:28.43 | [TK]D-Fender | <PROTECTED> |
| 19:28.59 | [TK]D-Fender | It could be sent inband and the call considered "active" because of that |
| 19:29.08 | [TK]D-Fender | we'd need to see full calls |
| 19:29.18 | [TK]D-Fender | Also you should confirm your exact version |
| 19:32.03 | sibiria | 13.18.2 right now, though i noticed the problem already on 13.13 which was the first version we moved to when upgrading these old 1.8 boxes |
| 19:32.18 | sibiria | i'm sure you nailed it with your inband theory |
| 19:35.38 | sibiria | the few (extremely few) occasions we see a difference between billsec and duration, it's billsec being 1 second longer than duration |
| 19:35.42 | sibiria | which is, uh, confusing to say the least |
| 19:40.16 | sibiria | we're also on chan_sip still.. *shrug* |
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| 20:43.55 | jcpeters01 | Worldexe, how was the domain pointing in /etc/hosts? |
| 20:45.58 | Worldexe | well, there is record like '1.2.3.4 mydomain' in /etc/hosts |
| 20:46.14 | Worldexe | so, 1.2.3.4 point to real peer ip |
| 20:47.27 | jcpeters01 | ok, thats what I would have guessed |
| 20:47.29 | jcpeters01 | thanks |
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| 23:34.11 | dl12 | Hey guys. Can asterisk run on hyperv without issues? |