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How to set up Asterisk 13 realtime sip properly without rtcachefriends=yes?
Good afternoon!
I'm trying to move from static sip.conf to realtime. It will be necessary to regularly make changes to the configuration, such as for example callerid. I created a plate, filled it out, set everything up in accordance with numerous manuals and set it up, as recommended in many
rtcachefriends=yes
rtupdate=yes
ignoreregexpire=yes
rtcachefriends=yes
I am not satisfied, because in order to update the information I need to either do sip prune realtime peer
orsip reload
, which leads to a temporary dump of devices until re-registration. I tried to put rtcachefriends=no
. INVITE sip:[email protected]:49795;rinstance=c7039f8233d45d0c SIP/2.0
Via: SIP/2.0/TCP 10.100.1.125:5060;branch=z9hG4bK07aa7564
Max-Forwards: 70
From: "6767" <sip:[email protected]>;tag=as588405ff
To: <sip:[email protected]:49795;rinstance=c7039f8233d45d0c>
Contact: <sip:[email protected]:5060;transport=TCP>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX GIT-13-11d05fc
Date: Mon, 16 Jan 2017 10:07:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281
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It looks like the RT mechanism is broken in my version of Asterisk. Even with rtcachefriends=yes, strange problems appeared in the system operation - calls from some devices did not occur through external trunks, despite the fact that the trunks were set to static. The behavior is similar to the one described - the package is indicated in the debug - in fact, there is no exchange of packages. There are no errors in the logs. There is no way to debug in production, I’m postponing the idea until the planned system update.
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