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| 00:17.54 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 05:51.00 | Gestahlt | Hi |
| 05:53.15 | Gestahlt | I have a strange problem. We have an asterisk installation and it works pretty good so far. Strange thing is: We tried to set up call forwarding to an external number (mobile). It "kinda" works. When we set progress inband to false, the call gets forwarded but there is no audio on pickup. If we set progress inband to true, we hear no ringtone indication, but on pickup, audio works just fine |
| 05:53.17 | Gestahlt | any ideas? |
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| 06:07.42 | Gestahlt | Ah this fixed it: https://community.freepbx.org/t/solved-no-sound-when-do-followme-on-ext-call-to-ext-number/29367/20 |
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| 09:07.28 | catphish | good day, i'd like some advice regarding channel event logging, i'm hoping to use it for simple billing by looking only at the CEL entries for channels relating to peers that bill me for calls |
| 09:09.18 | catphish | however i've run into an immediate problem, when a peer that represents an end user makes an outbound call using dial, it creates such a channel (to the upstream telco) but there is no reference to the remote username in the telco CEL entries, so i can't tell what number was ultimately called, and hence the per minute cost of the call |
| 09:10.56 | catphish | nor does it seemingly link in any way to the person who placed the call, so in CEL it's as if the second leg channel appeared out of nowhere |
| 09:16.10 | catphish | here's my complete CEL: https://paste.ubuntu.com/p/X8VWdWxW5V/ i guess "linkedid" will be my friend, or i could set CEL variables, but any advice on how to bill with this data would be appreciated |
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| 13:42.42 | Samot | catphish: CEL is for call events like transfer, etc. |
| 13:42.49 | Samot | It's not the best option for billing. |
| 13:50.34 | file | catphish: if you want to convey additional information - https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_CELGenUserEvent |
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| 13:58.05 | catphish | Samot: i was under the impression that CEL was designed to support billing, is there a better option within asterisk itself? |
| 13:58.40 | file | it can be used to do billing, and provides things to convey additional information if need be |
| 13:58.50 | file | it can't be used to do billing out of the box without a post processor |
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| 13:59.46 | catphish | file: thanks, i'll see if i can use CELGenUserEvent to add the data i need, the interesting problem is that i was hoping to bill based on usage of external (to an upstream carrier) channels, but because of the way dial() creates these, they seem to lack data |
| 14:00.46 | catphish | would i be better using "linkedid" and reassembling the whole call? essentially i just want to ensure that for every channel that connects to an external carrier, i want to work out who originated it and what that carrier might bill me |
| 14:01.15 | file | I haven't written a post processor, but linkedid is what can tell you the association |
| 14:01.30 | catphish | i think both of those will need me to use "linkedid" to join the channel back to whatever created it |
| 14:01.32 | catphish | thanks! |
| 14:02.56 | catphish | i can likely write a postprocessor to join each "external" channel to an originator (unless it is itself the originator, as would be the case for an incoming call) |
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| 15:29.59 | karelk | I need to add a new SIP provider account to my sip.conf |
| 15:30.10 | karelk | the SIP provider is using SRTP |
| 15:30.22 | karelk | where do I tell asterisk that the server is using SRTP ? |
| 15:30.36 | karelk | normally, the register command is: |
| 15:30.38 | karelk | register => user:secret@host/extension |
| 15:31.51 | file | it's configured in the SIP peer/friend/user, specifically the encryption option. |
| 15:36.32 | karelk | OK, so I have to add these ? |
| 15:36.33 | karelk | srtpcapable = yes, encryption = yes, forceencrypt = yes, transport = tls |
| 15:37.04 | file | only two of those options exist in chan_sip |
| 15:37.16 | file | encryption and transport |
| 15:37.32 | file | "encryption" enables SRTP as you originally asked about |
| 15:37.49 | file | if you also need to use TLS then more configuration is required, and is probably talked about in the sip.conf.sample file |
| 15:43.56 | karelk | I have TLS and SRTP already enabled and configured for other extensions |
| 15:44.12 | karelk | so I just needed to enable it for this new SIP provider |
| 15:44.45 | karelk | I added those 4 options and it seems to work |
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| 21:10.55 | FuriousGeorge | hey all |
| 21:11.23 | FuriousGeorge | if i want to goto dialstatus, then ${EXTEN} will go out of scope, no? |
| 21:23.22 | [TK]D-Fender | ${EXTEN} is where you are |
| 21:23.40 | [TK]D-Fender | When you Goto .. you're about to be somewhere else |
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| 21:30.41 | FuriousGeorge | [TK]D-Fender: thanks |
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