[asterisk-bugs] [JIRA] Commented: (ASTERISK-20333) 'directmedia' not working any more

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20333?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=197857#comment-197857 ] 

Matt Jordan commented on ASTERISK-20333:
----------------------------------------

I am completely unable to reproduce this problem using the latest from SVN 1.8.

Test Phone 1 (digium01):
{code}
[digium01]
type = friend
secret = digium01
directmedia = yes
callerid = "Digium 01" <101>
host = dynamic
disallow = all
allow = gsm
allow = ulaw
allow = alaw
allow = g722
context = from-digium-phones
registertrying = yes
{code}


Test Phone 2 (digium02):
{code}
[digium02]
type = friend
secret = digium02
directmedia = yes
callerid = "Digium 02" <102>
host = dynamic
disallow = all
allow = gsm
allow = ulaw
allow = alaw
allow = g722
context = from-digium-phones
registertrying = yes
{code}

Sample CLI:

{code}
[mjordan-desktop 1.8]$ asterisk -V
Asterisk SVN-branch-1.8-r374177
{code}

{code}
*CLI>   == Using SIP RTP CoS mark 5
    -- Executing [102 at from-digium-phones:1] NoOp("SIP/digium01-00000000", "") in new stack
    -- Executing [102 at from-digium-phones:2] Dial("SIP/digium01-00000000", "SIP/digium02") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/digium02
    -- SIP/digium02-00000001 is ringing
[Oct  2 14:26:57] NOTICE[25083]: chan_sip.c:25054 handle_request_subscribe: Received SIP subscribe for peer without mailbox: digium01
    -- SIP/digium02-00000001 answered SIP/digium01-00000000
    -- Remotely bridging SIP/digium01-00000000 and SIP/digium02-00000001
[Oct  2 14:27:01] NOTICE[25083]: chan_sip.c:25054 handle_request_subscribe: Received SIP subscribe for peer without mailbox: digium02
  == Spawn extension (from-digium-phones, 102, 2) exited non-zero on 'SIP/digium01-00000000'
{code}

Note that I used the peer configurations you provided in your sample sip.conf.  I'm not sure what the issue is, but without any further evidence of this issue persisting in Asterisk 1.8, I'm inclined to think it should be closed as "Can't Reproduce".

> 'directmedia' not working any more
> ----------------------------------
>
>                 Key: ASTERISK-20333
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20333
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.15.0
>            Reporter: Maciej Krajewski
>         Attachments: directmedia.cap, full, full+sipdebug.log, sip.conf, sippeers.csv, sippeers_propper.txt
>
>
> I've updated me asterisk today from 1.8.13.1 to 1.8.15.0,, what I've noticed is that directmedia has stopped working.
> I'm using realtime peer configuration, I have directmedia set to yes on both friends calling each other (simple dial without any options).
> {code}
> exten => 1,1,Dial(SIP/test002)
> same => n,Hangup()
> {code}
> Configured peers has dtmfmode set to "info" (I know this is an issue that direct media ins not working if set to rfc2833 or inbound).
> When I've downgraded asterisk to 1.8.13.1 the directmedia has worked again. I've done some investigation and what I  have found is that some change between 1.8.13.1 and 1.8.14.0 has done that.
> I've attached the logs from asterisk 1.8.15.0

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