[asterisk-bugs] [JIRA] (ASTERISK-27663) Register doesn’t work properly in Asterisk Realtime - not ringing calls
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Fri Feb 9 06:57:13 CST 2018
[ https://issues.asterisk.org/jira/browse/ASTERISK-27663?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242089#comment-242089 ]
Asterisk Team commented on ASTERISK-27663:
------------------------------------------
Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.
A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
> Register doesn’t work properly in Asterisk Realtime - not ringing calls
> -----------------------------------------------------------------------
>
> Key: ASTERISK-27663
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-27663
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/Registration
> Affects Versions: 13.18.5
> Environment: Ubuntu 16.04 amazon cloud. Clients behind the NAT
> Reporter: LEANDRO MORAES DE OLIVEIRA
> Severity: Critical
>
> I have an Asterisk server 13.18.5 installed in an Ubuntu 16.04 LTS in an Amazon cloud behind the NAT that was working properly, until I try to enable the Asterisk Realtime to use Mysql database.
> I made a lot of tests, in many situations work properly, but sometimes I lost the register of the client and the calls are not delivered, and in the server return it:
> [2018-02-09 09:32:23] WARNING[6142]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission 5aa7afc34fa9290b5b4967a966462a91 at 54.94.200.147:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> I already use rtpcachefriends=yes and rtsavesysname=yes. When I didn’t use RealTime I solved it with register=>4701006:password at 10.120.23.24/4701006 now when I use this command the calls was not delivered and say that the phone is in loop detected:
> -- Got SIP response 482 Loop Detected back from IP_OF_MY_SERVER
> -- SIP/BuyDIDHub-0000069e is circuit-busy
> == Everyone is busy/congested at this time (1:0/1/0)
> And then the call down.
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