IRC log for #asterisk on 20190603

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08:14.32alexandre9099hi, so i have a reaaly old computer which seems to recognize it's rj11 port a sound device, would it be possible to use it as a trunk(?)
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08:36.32ApocryphalJust upgraded to Asterisk 16 with PJSIP, and there is this new thing with blind transfers. Probably PJSIP being better than chan_sip was, but if I do a blind transfer, in this case directly to a users voicemail, and if the transferee hangs up during the voicemail greeting, then Asterisk sends a NOTIFY with 503 Service Unavailable to the transferer. Depending a bit on the phone model, this cause the
08:36.38Apocryphalphone to do a "Redial" (presumably to attempt to reconnect to the transferee)
08:36.46ApocryphalCan that be controlled through some sort of settings?
08:37.05ApocryphalMainly I'd like asterisk to not send out that 503 at all if possible
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08:43.21ApocryphalOkay can be controlled with refer_blind_progress on the pjsip endpoint.
08:44.11ApocryphalI'm curious what will cause asterisk to send the 503 message out though. In this case it seems to happen if the transferee hangs up while listening audio playing by the STREAMFILE (AGI command)
08:44.29ApocryphalBut if they hang up inside an ARI app it'll send 200 just fine
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14:09.25jkrooni've got a string (containing json encoded stuff, to be sent to the telegram API).
14:09.56jkroonThen I do NoOP(${CURL(https://api.telegram.org/bot754924937:AAEmC-F0vhVnaQjISR4xbrvNjNPiRmANbdY/sendMessage,${LOCAL(post)})}))
14:10.15jkrooneesh, ok, so that's a new bot token for me :).
14:11.32jkroonthe problem I find is that the , in the json encoded data is causing a third argument to be passed (I think) which ends up getting ignored.
14:16.43[TK]D-Fenderoption: use an AGI for this instead and a native CURL or shell call instead of the dialplan function.
14:18.38avbjkroon: you can try to use encoded version of ,
14:19.10avb%2C
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14:19.53avbbut yes, some external script sounds more manageble
14:20.05jkroonavb, that'll go into the POST data as %2C ... so I suspect won't work.  I have however based on that idea switched to not using json and instead url encoding the data ... which seems to be better at this.
14:20.37avbPOST is always better, right
14:20.45avbsince GET have size limitation
14:21.03jkroonavb, the issue is CURL(url[,post]) ... so CURL(url,post,with,a,comma) ends up just posting post.
14:21.24avbah
14:21.58avbNoOP(${CURL(https://api.telegram.org/bot754924937:AAEmC-F0vhVnaQjISR4xbrvNjNPiRmANbdY/sendMessage,${STR_REPLACE(${LOCAL(post)},",","%2C"))}))
14:22.01avb:)
14:22.14avbim not sure that i have used a correct synta
14:22.15avbim not sure that i have used a correct syntax
14:22.22avbbut you got an idea
14:22.49avbhttps://www.voip-info.org/asterisk-func-strreplace/
14:23.05jkroonSTR_REPLACE(LOCAL(post),",","%2C") ... interesting, assuming that the quotes doesn't get passed as part of the argument, as per my experience.
14:23.21jkroonhttps://wiki.asterisk.org/wiki/display/AST/Parameter+Quoting
14:23.44avbas well
14:24.06avbim using quote without thinking about it, funny
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14:27.04jkroonyea, ok, so STR_REPLACE is quite a good option to have available.  However:
14:27.06jkroon<PROTECTED>
14:27.07jkroon<PROTECTED>
14:27.52jkroonso basically, Set(LOCAL(post)=text=${URLENCODE(${LOCAL(text)})}) ... and then you post that.
14:28.08jkroonsince , will be encoded, no more nasty argument issues, and quotes too should get encoded.
14:30.07avbthere you go
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15:47.07cuscohello
15:47.56cuscoI'm compiling asterisk 16 for the first time.. I used configure with bundled jansson and bundled pjproject...
15:48.42cuscoI'm getting: Error loading module 'res_pjsip_config_wizard.so': /usr/lib/asterisk/modules/res_pjsip_config_wizard.so: undefined symbol: ast_sip_get_sorcery
15:48.47cuscocan't figure why
15:49.23cuscogoogling shows something like upstream pjproject 2.5 has changed its API and should used 2.4.5 instead ..
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17:04.37cuscohmm.. for the init script to detatch, I need -F ?
17:05.29cuscoI'm getting: 'alwaysfork' is not compatible with console or remote console mode; ignored
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19:57.42mmx87Hello, I'm getting an error, but I'm not sure what I'm doing wrong.
19:57.42mmx87The error is: ERROR[6954][C-00000000] app_stack.c: Attempt to reach a non-existent destination for Gosub: (Context:sub, Extension: s, Priority:1)
19:57.42mmx87https://0x0.st/zMUr.txt
19:57.42mmx87Could anyone tell me why it can't "find" that context?
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20:11.18rmudgettmmx87:  Remove those spaces after commas.  It is trying to find extension ' s' when you have extension 's'.
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20:13.24seanbrighti thought we fixed that
20:14.22rmudgettDialplan parsing is very inconsistent where spaces are concerned.
20:15.23rmudgettMainly because parameter parsing is done by each application.
20:15.55seanbrightthe goto apps use main.c:pbx_parseable_goto
20:16.12seanbrightbut it does not space skipping, so...
20:17.16mmx87Oh, I see now. It was because of the space before the s....
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20:36.20mmx87Another question. TLS works for me but TCP transport doesn't. At least I'm unable to connect on a mobile network.
20:36.21mmx87Is there something that makes SIP less likely to "fail" when used with TLS?
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