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| 00:21.54 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); 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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| 00:23.10 | Bhakimi | is there a issue with the chat ? |
| 00:23.16 | Bhakimi | i keep getting kicked out |
| 00:23.36 | Bhakimi | i was wondering if there is a way to control the volume of an exsisting channel |
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| 01:51.26 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); 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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| 09:32.57 | gvs77 | Hi, is it possible to have Asterisk redirect/transfer a call to an external number (GSM) after a while? With redirect I mean release control of the call to the other number (which keeps callerid intact) |
| 09:33.09 | gvs77 | I'm using a SIP trunk BTW |
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| 12:24.29 | [TK]D-Fender | gvs77, What does "after a while" mean? When would it make this decision? Why? |
| 12:25.22 | [TK]D-Fender | gvs77, If you want to take a channel and send it out then it's your dialplan. You'll have to defin where that happens |
| 12:29.32 | gvs77 | [TK]D-Fender, I just need a method to transfer to external with callerid intact, I can set a remote destination but it does not preserve callerid |
| 12:30.03 | [TK]D-Fender | That depends on how yo call out |
| 12:30.11 | [TK]D-Fender | go look at that call and your peer config |
| 12:30.27 | [TK]D-Fender | not all providers allow you to set the callerid |
| 12:30.33 | [TK]D-Fender | so go prove what you are sending them |
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| 12:39.38 | gvs77 | [TK]D-Fender, Most SIP providers here don't, mine doesn't. So someone mentioned that it could be possible to transfer the call, which means loosing control over it |
| 12:41.01 | [TK]D-Fender | show the call to prove what you're sending |
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| 12:42.04 | gvs77 | [TK]D-Fender, That's what I'm asking, if there is a way to send a redirect/ transfer like I describe. I asked my provider, I can't send arbitrary callerid. |
| 12:42.21 | [TK]D-Fender | then the reason you can't is your PROVIDER |
| 12:42.33 | [TK]D-Fender | so get another one |
| 12:43.05 | gvs77 | [TK]D-Fender, All providers in Belgium that I know don't allow it |
| 12:43.15 | [TK]D-Fender | then get one outside |
| 12:43.32 | gvs77 | [TK]D-Fender, I need to keep my current local dial-in number |
| 12:43.41 | [TK]D-Fender | ... |
| 12:43.46 | [TK]D-Fender | get another one to dial this OUT <- |
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| 12:43.59 | [TK]D-Fender | You can have 100 providers if you want or need |
| 12:44.57 | [TK]D-Fender | Your current provider does not let you set an arbitrary CID. There is no way to FORCE them. use a service that DOES. Kepp your original for the things you need them for and use another provider for your other needs |
| 12:46.24 | gvs77 | [TK]D-Fender, I just wanted to see if there was another method which someone mentioned to redirect the call to the external number. It's not worth the added cost and complexity of having another SIP provider |
| 12:46.37 | [TK]D-Fender | that means nothing |
| 12:46.42 | [TK]D-Fender | you are sending the call OUT a provider |
| 12:46.46 | [TK]D-Fender | YOURS doesn't let you |
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| 12:47.04 | [TK]D-Fender | There is no miracle way your call is going out without a SERVICE being provided to you that lets you do it |
| 12:47.07 | [TK]D-Fender | and yours doesn't |
| 12:47.10 | gvs77 | [TK]D-Fender, hence I looked for something similar to the http permanent redirect |
| 12:47.25 | sekil | gvs77: you mean like isdn deflect.. |
| 12:47.27 | [TK]D-Fender | you are failing to grasp that some PROVIDER is accepting that call |
| 12:47.31 | [TK]D-Fender | they are clearly giving you SERVICE |
| 12:47.35 | sekil | gvs77: yeah many SIP providers don't allow this.. |
| 12:47.41 | [TK]D-Fender | YOURS doesn't do what you want |
| 12:47.44 | [TK]D-Fender | you'll need ANOTHER |
| 12:47.47 | gvs77 | sekil, that is what I'm looking for |
| 12:47.58 | [TK]D-Fender | Pick another provider for that need |
| 12:49.09 | gvs77 | sekil, thanks for the pointer, I found this: http://developer.beronet.com/index.php/Howto_use_Calldeflection/Partial_Rerouting_with_Asterisk_based_systems |
| 12:49.12 | sekil | gvs77: you can try to add some sip headers on the outbound invite...here we need to add Diversion header |
| 12:50.11 | sekil | gvs77: yeah but like [TK]D-Fender this will not work.. |
| 12:50.26 | sekil | gvs77: only i.e. if they react to say Diversion header populated |
| 12:50.27 | gvs77 | sekil, because the trunk blocks it? |
| 12:50.57 | sekil | gvs77: usually providers have a default clid to specify if it's not the one from the allowed range |
| 12:51.11 | sekil | gvs77: like a "leading" clid to bill users with |
| 12:51.49 | sekil | gvs77: so if you're sending an original clid it will be rewritten |
| 12:52.45 | sekil | gvs77: there's not much you can do on your end...in fact nothing if they don't allow it |
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| 12:53.19 | gvs77 | sekil, also good to know... It's not worth the additional cost to me, so I will work around it |
| 12:54.29 | sekil | gvs77: isdn deflect was a very good service in the "old" days..:) |
| 12:54.57 | [TK]D-Fender | because your ISDN telco permitted it as well. |
| 12:55.00 | gvs77 | sekil, yes, that's why I thought it would be possible. I'd rather have Asterisk manage the call end to end |
| 12:55.06 | [TK]D-Fender | provider still has to support these things |
| 12:55.08 | sekil | [TK]D-Fender: exactly |
| 12:55.49 | gvs77 | Belgian providers aren't the greatest unfortunately... |
| 12:56.14 | sekil | gvs77: maybe there's something like **95<forwarding number> |
| 12:56.18 | [TK]D-Fender | <[TK]D-Fender> Pick another provider for that need |
| 12:56.33 | [TK]D-Fender | <[TK]D-Fender> get another one to dial this OUT <- |
| 12:56.42 | [TK]D-Fender | <[TK]D-Fender> so get another one |
| 12:57.41 | gvs77 | [TK]D-Fender, I agree if it is worth the cost, I need to still get a EU provider for billing and not worth the cost |
| 12:58.14 | [TK]D-Fender | Then this idea pretty much stops right here |
| 12:59.26 | gvs77 | [TK]D-Fender, yes, like I said, not worth the cost so I will work around it |
| 12:59.57 | sekil | gvs77: what do you consider work around here, I'm curious? |
| 13:00.10 | sekil | gvs77: as a work around that is |
| 13:00.29 | gvs77 | Probably announced pick up on the cellphone |
| 13:00.48 | gvs77 | sekil, downside is that it adds to the call duration and the caller will hang up |
| 13:01.03 | sekil | gvs77: that's really bad |
| 13:01.25 | [TK]D-Fender | guess the calls can't be that important then |
| 13:01.54 | gvs77 | [TK]D-Fender, not worth the cost |
| 13:02.20 | gvs77 | And another option is to keep the cellphone on VPN and registered locally |
| 13:02.37 | [TK]D-Fender | Yup, so you've got your other method of trying to convey that information and callers may hang up. I guess that's what you've got left. |
| 13:04.40 | gvs77 | [TK]D-Fender, that's life :-) |
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| 13:46.27 | skrusty | Odd question, but... is there any way with pjsip to provide sip creds at the point of making a call? |
| 13:48.47 | newtonr | You mean before the 401 unauthorized? |
| 13:49.14 | newtonr | I dunno |
| 13:49.26 | skrusty | yeah |
| 13:49.50 | skrusty | i need to be able to route a call to somewhere, and provide asterisk with the u/p in the event of auth request |
| 13:57.15 | file | there isn't |
| 13:57.15 | file | not as of this time |
| 13:57.15 | Samot | Nope. |
| 13:57.15 | Samot | Kamailio.. |
| 13:59.32 | skrusty | :'( |
| 13:59.35 | skrusty | yeah |
| 13:59.48 | skrusty | i was trying to simplify things though |
| 13:59.56 | skrusty | without having to use Kamailio to do something so simple |
| 14:00.02 | Samot | You're going to have to modify the SIP message |
| 14:00.41 | Samot | Is the other side actually sending back a 401? |
| 14:00.51 | skrusty | yeah |
| 14:01.07 | Samot | But Asterisk isn't sending back auth digest? |
| 14:01.20 | skrusty | because it has no auth in the db |
| 14:01.34 | skrusty | i don't want this to be stored anywhere - does that make sense? |
| 14:01.45 | Samot | Sure. |
| 14:01.52 | skrusty | almost like i want an `at runtime` auth that i can just pass along |
| 14:01.54 | Samot | You'll have to use a SIP router. |
| 14:01.58 | skrusty | yeah :( |
| 14:02.18 | Samot | You're going to have to interject changes to the INVITE you send back |
| 14:02.47 | skrusty | i've already done it in kamailio |
| 14:02.51 | skrusty | using the auc module |
| 14:02.54 | skrusty | uac even |
| 14:03.00 | Samot | Yup. |
| 14:03.10 | skrusty | i am trying to remove complications though :) |
| 14:03.26 | Samot | Trying to do it from Asterisk would just add them |
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| 15:25.28 | igcewieling | file, you around? |
| 15:31.38 | file | Barely |
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| 16:16.55 | igcewieling | Is this something which is expected to work? Set(MASTER_CHANNEL(__TESTVAR)=1234) i.e. starting at the top channel, set TESTVAR on all child channels? |
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| 18:24.46 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); 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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| 19:43.07 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); 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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