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| 12:58.28 | [sID] | Hello, |
| 12:58.32 | [sID] | The question is, can I somehow get information about REFERRED-BY: and Refer-To: in dialplan? |
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| 14:02.53 | sibiria | [sID]: you can read headers using the PJSIP_HEADER() function |
| 14:03.10 | sibiria | for chan_sip i believe the function is SIP_HEADER() |
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| 14:04.29 | [sID] | sibiria: |
| 14:04.31 | [sID] | Well, according to the documentation, the header itself does not provide information. |
| 14:04.34 | [sID] | https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_SIP_HEADER |
| 14:04.34 | [sID] | I don't understand how to use the GET_TRANSFERRER_DATA variable |
| 14:04.54 | [sID] | exten => _.,1,Set(TRANSFER=${SIP_HEADER(GET_TRANSFERRER_DATA(x-.))}) |
| 14:05.11 | [sID] | Yes unfortunately it does not download any information. |
| 14:07.45 | sibiria | "The variable must be set before a call to the application that starts the channel that may eventually transfer back into the dialplan, and must be inherited by that channel, so prefix it with the _ or __ when setting (or set it in the pre-dial handler executed on the new channel)." |
| 14:07.50 | sibiria | did you do this as well |
| 14:09.49 | sibiria | i would give SET(GET_TRANSFERRER_DATA=x-) a try, before initiating the call |
| 14:09.55 | [sID] | Well, at the beginning I have dialplans of this context set up |
| 14:10.20 | sibiria | if you originate with a call file, you must set it there instead i believe |
| 14:10.49 | sibiria | since once it does enter the context that way, that channel is already bridged |
| 14:11.58 | [sID] | Ok, check it out for a moment |
| 14:13.42 | [sID] | sibiria: Ok, check it out for a moment |
| 14:13.49 | [sID] | He set up, now the question is how to download it exactly? |
| 14:14.12 | Samot | ? |
| 14:14.55 | sibiria | i'm not actually sure, but from this point i would try just accessing the header directly by its name with SIP_HEADER() |
| 14:15.42 | [sID] | Ultimately, I need to get information Refer-To: and REFERRED-BY |
| 14:15.52 | sibiria | ah sorry |
| 14:15.53 | sibiria | no |
| 14:16.06 | sibiria | i didn't read the last part - headers will be returned to your channel as a hash |
| 14:16.15 | sibiria | "Headers are returned in the form of a dialplan hash TRANSFER_DATA, and can be accessed with the functions HASHKEYS(TRANSFER_DATA) and, e. g., HASH(TRANSFER_DATA,X-That-Special-Header)." |
| 14:16.34 | sibiria | so the latter variable call is what you're looking for |
| 14:17.34 | sibiria | e.g. Verbose(1,${HASH(TRANSFER_DATA,SOMEHEADERNAME)}) to have a look |
| 14:18.13 | sibiria | i get the feeling this would be easier with pjsip :P |
| 14:18.28 | sibiria | maybe consider it an incentive to finally move along from chan_sip |
| 14:19.35 | [sID] | It turns out :) I'll check in a moment. |
| 14:20.28 | [sID] | no return :( |
| 14:21.15 | [sID] | Executing [192.168.1.50:5061@from_sbc:1] Set("SIP/customer-cx_001-00000152", "GET_TRANSFERRER_DATA=x-") in new stack |
| 14:21.33 | [sID] | Executing [192.168.1.50:5061@from_sbc:9] Verbose("SIP/customer-cx_001-00000152", "1,") in new stack |
| 14:21.42 | Samot | [sID]: Do you have hedaders that stat with x-? |
| 14:21.55 | Samot | [sID]: Do you have headers that start with x-? |
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| 14:22.28 | jaysbnc | hi guys, anyone can support me connecting to companyflex sip trunk via pjsip? |
| 14:24.54 | [sID] | https://pastebin.com/rUpKVhig |
| 14:25.08 | [sID] | This Header |
| 14:26.15 | Samot | And what is the dialplan for this? |
| 14:28.50 | [sID] | https://pastebin.com/RFeqP2T7 |
| 14:30.42 | [sID] | hmm |
| 14:30.44 | [sID] | its work |
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| 14:31.31 | [sID] | Something works, I'm still checking |
| 14:38.09 | jaysbnc | hey guys, anyone familiar with exotic sip trunks? |
| 14:38.23 | jaysbnc | i am strugling with an companyflex account |
| 14:40.42 | sibiria | no, but if you provide some more details maybe someone has an idea |
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| 19:19.21 | igcewieling | whoops |
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| 20:20.24 | drmessano | Yeah, you better whoops |
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| 21:55.27 | velix | Sorry for asking, but can I assign variables anywhere or only in extensions? f.e. sounds => '/other/location/' |
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| 22:03.10 | Samot | Just in diol |
| 22:03.16 | Samot | Dialplan |
| 22:03.49 | Samot | You can set vars on pjsip or sip channels via the endpoint/peers config |
| 22:04.59 | velix | Samot: Oh okay. Then I need to link the directory. |
| 22:05.56 | Samot | Link what directory and how? |
| 22:07.49 | velix | Before answering this (it's complicated), can the directory variables (f.e. astetcdir) set in asterisk.conf be read from other configs or the dialplan? |
| 22:10.06 | Samot | velix: What don't you explain what you are trying to do instead of asking questions one at a time. |
| 22:13.05 | velix | Samot: It's always the same. When you'd simply answer the questions, all would be good. But you want to supervise my setup... |
| 22:13.06 | velix | https://bpa.st/raw/ZY6A |
| 22:13.12 | velix | I've explained it for you in detail. |
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| 22:13.46 | velix | The first thing when I come to this channel is to apologize for aksing a question... |
| 22:14.23 | Samot | OK, nevermind. |
| 22:14.33 | Samot | You asked a question, I gave an answer. |
| 22:14.53 | velix | Okay, thanks for your time. |
| 22:14.58 | Samot | You asked another question, I asked for clarification on that question, you decided to not answer but ask a different question. |
| 22:14.58 | velix | (and for the answer) |
| 22:16.57 | velix | My question was about using variables in a path, f.e. "$astdatadir/sounds/myownsound.gsm" in dialplan. |
| 22:17.38 | Samot | Of course you can use vars in the path. |
| 22:17.55 | velix | but they need to be defined in the callfile. |
| 22:18.05 | velix | I mean, I cannot access the variable set in asterisk.conf? |
| 22:20.43 | velix | ah, it works. ${ASTDATADIR} |
| 22:20.45 | velix | Thanks! |
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| 22:28.31 | velix | Sorry for asking another question: can I keep the DB from 17.8 to 18.1 or shall I recreate it? |
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| 22:38.13 | velix | Samot: The symlink solution also works. I've got two approaches now. Thanks for your knowledge again. |
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| 23:26.03 | aNullValue | does anyone here have experience with devices that do video calling over SIP, who might be willing to give advice? |
| 23:27.35 | aNullValue | i need to implement a video door intercom system in a factory. probably with an Axis video door station, which speaks SIP. i've seen a few videos of it being done on youtube, so it works in theory at least, but... i'm concerned about the phones (on the inside, not the door phone), how long they're going to last, how complex the user experience is, etc |
| 23:27.58 | aNullValue | mostly because the only sip phones i can find that seem to support video are running downright ancient versions of android |
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| 23:35.32 | Samot | aNullValue: Poly's support video. Others support video but you need to get mid-high range models of their phones. |
| 23:35.53 | Samot | Because either the phone requires an external camera via USB or has one built in. |
| 23:40.38 | aNullValue | all polys? i've been looking at spec sheets and not seeing video codec support :-\ |
| 23:40.45 | aNullValue | maybe i'm missing something, heh |
| 23:42.24 | aNullValue | i really don't need the desk phones to have a camera, i just need them to display the video from the far side. heh. |
| 23:57.20 | Samot | My points was simple. |
| 23:57.32 | Samot | There are numerous SIP phones that support this. |
| 23:57.43 | Samot | Poly, Yealink, Grandstream... |
| 23:58.03 | Samot | Three big players in the space. All have a model or more that support video in some format. |