Archives : June-2020
after migration from chan_sip to res_pjsip I get strange behavior when receiving call from the outside world. When call is received, it is replicated multiple times. Two of that calls get to the phone. So the phone is ringing on both lines. When hav..
–0000000000000d207405a7581d19Content-Type: text/plain; charset=UTF-8Greetings All,Weve been working hard on new codec negotiation stuff for Asterisk 18 and weve got some stuff to run by you.Its a lot so please read carefully.To give you some idea..
I would like to ask about current state of subscribecontext in pjsip. I found out some 6 years old discussion on that without any plans to implement it in the future. I have phones in different contexts. I suspect, when I use its context to subscri..
!Is there something wrong with the function HINT(,n) ?Note the param n – it is supposed to get (as far as I understand the documentation) CALLERID(name) of the extension (via extensions hint).Example configuration:pjsip.conf:[10]type=endopoint.. calle..
Please see the above footer, as shown on all list emails. Regards, Antony. — Pavlov is in the pub enjoying a pint. The barman rings for last orders, and Pavlov jumps up exclaiming Damn!I forgot to feed the dog! Please reply to the list; please *do..
I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there. I ran into an issue when testing that incoming calls from MagicJack would go sil..
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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