Archives : November-2020
i want gradually migrate to pjsip i have 2 network interfaces with a lot of endpoints on both (both UDP) eth0 1.1.1.1 eth1 2.2.2.2 i want start migrate endpoints to pjsip on 2.2.2.2:5070 but configuration like this is not possible sip.conf [gener..
Is anyone aware of any way of changing the contact header on a call? We are sending 911 calls to a provider and they require that the contact be the call back number. I tried:Set(PJSIP_HEADER(update,contact)=)But the came back with:No headers had b..
What is the most FHS-esque (see [1])way to run several Asterisk instances on a single (Debian) host ?What would you recommend ?Would gather each instance directories (etc/, run/, lib/, …)in something like /srv/instance1/(it doesnt please me as I l..
Im working on converting my 18.0.1 test system from SIP to PJSIP and Ive run into something odd.I have a queue defined named acme-test that has two agents in it, PJSIP/7001acme and PJSIP/7002acme.I have autohints=yes in my acme-intern context, I h..
The Asterisk Development Team would like to announce the release of Asterisk 18.1.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 18.1.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 17.9.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 17.9.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 16.15.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 16.15.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 13.38.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 13.38.0 resolves several issues repor..
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..
i try to connect my SIP Client (linphone) via VPN to FreePBX. The routing looks OK. I can ping the Endpoints and traffic is routing. I can also Register my Sip Client. debpbx*CLI> pjsip list contacts Contact: =======================================================================================..