can you recommend way to test status of PJSIP endpoint (SIP trunk to the operator)?
is there something better than parsing
asterisk -rx pjsip show contact operator/sip:operator@1.1.1.1:5060
?
We are using icinga2/prometheus
M..
Author : marek
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its possibe to dont start music on hold when caller (from sip operator trunk) press HOLD (i.e. on mobile phone)
Asterisk acts on SDP a=sendonly
i want pass trough media from SIP trunk provider
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patches welcome PhilipAsterisk is Open Source so everyone can helpMarekDne 12/05/2020 v 07:02 Saint Michael ..
any plans for astricon videos?
Thanks
M..
i have following topology PSTN – Asterisk —- internet —– router – jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router – public IP/private IP (NAT) jssip client – private IP – ..
i need call time of userB after attended transfer scenario 1) call from Customer to userA 2) userA start consultancy to userB (attended transfer started) 3) userA attended transfer to userB (transfer after consultacy) 4) userA hangup in CEL i have eventt..
If you are using JavaScript for *AGI/ARI/AMI we made small library for asterisk dialplan pattern matching and number manipulation https://www.npmjs.com/package/asterisk-pattern-matchingexamplesconst { validateNumber } = require(asterisk-pattern-matching..