[asterisk-bugs] [JIRA] (ASTERISK-24657) directmedia=no does not work in Asterisk 13.1.0
Thomas B. Clark (JIRA)
noreply at issues.asterisk.org
Fri Jan 2 21:17:34 CST 2015
Thomas B. Clark created ASTERISK-24657:
------------------------------------------
Summary: directmedia=no does not work in Asterisk 13.1.0
Key: ASTERISK-24657
URL: https://issues.asterisk.org/jira/browse/ASTERISK-24657
Project: Asterisk
Issue Type: Bug
Security Level: None
Affects Versions: 13.1.0
Environment: The Asterisk server runs under Fedora 20, with static ipv4 addresses and working ipv6.
The yealinkphone is a sip client of the asterisk server. The yealink phone has a non-routable ipv4 address and a routable ipv6 address, but preferentially uses the ipv6 address.
The service provider, callwithus, does not have an ipv6 address.
Reporter: Thomas B. Clark
I just upgraded from Asterisk 11.14.2 to asterisk 13.1.0, and that caused directmedia=no to stop working. It worked in 11.14, and using the same sip.conf it does not work in 13.1.0. Here is the relevant part of sip.conf:
[yealinkphone]
type=friend
host=dynamic
context=outgoing
secret=xxxxxx
defaultuser=xxxxxxx
insecure=port,invite
directmedia=no
qualify=yes
And here is what happens:
-- SIP/callwithus-00000003 is making progress passing it to SIP/yealinkphone-00000002
> 0x7fa784016470 -- Probation passed - setting RTP source address to 69.85.185.222:21708
-- SIP/callwithus-00000003 answered SIP/yealinkphone-00000002
-- Channel SIP/yealinkphone-00000002 joined 'simple_bridge' basic-bridge <5fb81915-9eaa-4b69-9be8-26f8a1db1fff>
-- Channel SIP/callwithus-00000003 joined 'simple_bridge' basic-bridge <5fb81915-9eaa-4b69-9be8-26f8a1db1fff>
> Bridge 5fb81915-9eaa-4b69-9be8-26f8a1db1fff: switching from simple_bridge technology to native_rtp
> 0x7fa70400afd0 -- Probation passed - setting RTP source address to [2601:8:9181:4800:215:65ff:fe27:ac8e]:11786
Even if I had directmedia=yes, Asterisk shouldn't issue a reinvite because the yealinkphone is using ipv6 and the service provider, callwithus, is using ipv4. However, as I stated, I have directmedia=no, so nothing else should matter, but the statement is being ignored.
I have worked around the issue by setting ,,t at the end of all of the Dial statements, and that successfully prevents switching to native_rtp. Nothing else I tried does.
--
This message was sent by Atlassian JIRA
(v6.2#6252)
More information about the asterisk-bugs
mailing list