[asterisk-bugs] [JIRA] (ASTERISK-26313) chan_sip : Asterisk restart seems to be required for changing encryption option

[Vitezslav Novy (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26313?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235442#comment-235442 ] 

Vitezslav Novy edited comment on ASTERISK-26313 at 2/26/17 7:26 AM:
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This problem seems to be fixed in asterisk 14. I attach patch for asterisk 13.13.1


was (Author: vnovy):
This problem seems to fixed in asterisk 14. I attach patch for asterisk 13.13.1

> chan_sip : Asterisk restart seems to be required for changing encryption option
> -------------------------------------------------------------------------------
>
>                 Key: ASTERISK-26313
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26313
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 13.10.0
>         Environment: Debian 8 3.16.0-4-amd64
>            Reporter: benasse
>            Assignee: Unassigned
>            Severity: Minor
>         Attachments: asterisk-13.13.1-dtls-reload.patch, full.log, sip1.conf, sip2.conf
>
>
> Below, the steps to reproduce :
> * I have a working peer, configured as in sip1.conf, I call the number 123
> * I remove the parameters related to WebRTC as in sip2.conf
> * I do a SIP reload
> * When I try to make a call I get the following message in the asterisk CLI :
> {code}WARNING[5107][C-0000000e] chan_sip.c: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio{code}
> Atached the asterisk log :
> First call:
> callid asterisk: C-0000000c
> CallID SIP: 83deae9a-5991-3a37-7c58-12f89d3ede6d
> Second call:
> callid asterisk: C-0000000e
> CallID SIP: d9c74c18-a685-418b-a8f5-ea9938ac3be3
> After a restart of asterisk, the line is working properly.
> Do not hesitate if you want more information.



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