since ipv6 doesnt really push ipv4 out of our networks, we still use IAX (when possible) to not face NAT problems from SIP, Asterisk 16 included. Question is, what is the future of IAX. Is it a dead protocol ? Will it stay as is even only security probl..
Author : Administrator TOOTAI
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I rey to register an Asterisk 16.2.1 pjsip to an ASTERISK 13.25.0 chan_sip using ipv6 and pjsip_wizard. I only got it work if in remote_hosts I put the ipv6 address and not the hostname like sip.domain.ltd No need to say that an AAAA entry is exist..
We upgraded an Asterisk 11 server to 16.1.1, going from chan_sip to pjsip, on a site using Gigaset phones. They are registring well despite the fact that we get a lot of errors like [Feb 22 18:30:07] ERROR[1556]: pjproject: :sip_transport. Error process..
Le 30/01/2019 à 05:17, Jose Tavares a écrit :
chan_ooh323 does not do the job ?
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I have an external agent which register dynamically in a queue and I setup his PJSIP account on a identity of a local phone which is configured to redirect all calls to the agent. Agent can have only on call at a time. When agent takes a call he is pau..
on an Asterisk 16 with PJSIP I want to know the state of a device (idle, busy, unavailable, …) in the dialplan. I tried with ChanIsAvail() but this one doesnt return the real state (eg a device calling an extension which is running ChanIsAvail()..
Im used to set call-limit in sip.conf Now I switched one customer Asterisk to 16 version and cant get the behavior back, as well for extensions as for queues. I set ringinuse=no for queues and have max_audio_streams = 1 max_video_streams = 0. I wan..
, to manage an external queue agent the only solution I found is to connect a local account and redirect calls to this account using forward features from the phone (SNOM). The problem I face is that before calling the agent I would like to set ex..
all, I want to set dynamic queue with non local members. I create an extension 115 in [localEP] context which is doing the job, eg calls to this extension are forwarded to the non local endpoint (which is an IP phone connected to an external Aster..
Im on the way to upgrade a dialplan from 1.8 to 16.0.1 and face a problem with user variable defined in sip.conf using setvar. It work like a charm -even on asterisk 13 version- but cant get it work in 16. The variables are defined in pjsip with set_..