im trying asterisk AEAP through Haproxy https://docs.asterisk.org/Asterisk_18_Documentation/API_Documentation/Module_Configuration/res_aeap/?h= backend speech-gateway-dev-wss mode http option forwardfor option http-server-cl..
Author : marek
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we are moving our asterisk from chan_sip to chan_pjsip we are using SIP_HEADER with GET_TRANSFERRER_DATA for headers from REFER (asterisk – other pbbx – SIP REFER – asterisk) https://github.com/ca4ti/asterisk/commit/4b58609c331c013845a0a61d946cbbc820921..
anybody using AEAP in production? https://www.asterisk.org/asterisk-external-application-protocol-speech-to-text-engine/ im trying, but i have problems with opus (no response from google) another problem is i need alaw and it looks like only ulaw..
there are new versions of Asterisk but mailing list is empty
http://lists.digium.com/pipermail/asterisk-users/
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in release notes for RHEL 9.1 i see — https://access.redhat.com/documentation/en-us/red_hat_enterprise_linux/9/html/9.1_release_notes/technology_previews KTLS available as a Technology Preview RHEL provides Kernel Transport Layer Security (KTLS)..
we are migrating from chan_sip to pjsip
i want logs like this about pjsip endpoints
[Oct 14 17:20:36] NOTICE[35629] chan_sip.c: Peer endpoint22 is now Reachable. (15ms / 2000ms)
is it possible?
thanks
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whats best method from perfomance view to parse this headerP-Asserted-Identity:P-Asserted-Identity:i need+44111222333 (if Privacy: id)PJSIP_PARSE_URI ?STRREPLACE/CUT?FastAGI?other options?th..
i have two asterisk boxes need transfer call from second box to first one pstn -> asterisk1 -> dial(number 555) -> asterisk2 -> TRANSFER (number 444) -> asterisk1 dialplan on asterisk1 (using chan_sip) [some_context] exten => 555,1,Noop() same => n,dial(SIP/asterisk2/5..
i have 2 queues – queue1 – queue2 1 agent is in both queues queue strategy is rrmemory i have 2 calls waiting call from 12:00 in queue1 from number 777 call from 12:05 in queue2 from number 666 at 12:10 agent is free for next call i have problem in t..
is it possible to playback ogg/opus files to alaw sip clients?exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output.ogg,)[Dec 22 14:05:53] WARNING[49275][C-00000004]: file.c:789 ast_openstream_full: File /var/lib/asterisk/sounds/output.ogg d..