[asterisk-bugs] [Asterisk 0016627]: No audio is passed from MOH when using originate to a remote peer

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=16627 
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Reported By:                kobaz
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16627
Category:                   Resources/res_musiconhold
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                       SWP-743 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 240716 
Request Review:              
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Date Submitted:             2010-01-17 14:18 CST
Last Modified:              2010-06-02 13:56 CDT
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Summary:                    No audio is passed from MOH when using originate to
a remote peer
Description: 
No audio is passed from MOH when using originate to a remote peer

If you issue a Playback(), before MusicOnHold, audio will flow.

---------

extend context services {
  1 => {
    Answer(500);
    MusicOnHold();
  }

  2 => {
    Answer(500);
    Playback(lunch);
    MusicOnHold();
  }
}

5506 is a polycom phone registered on pbx A
on pbx A:
  asterisk -rx "originate SIP/5506 extension 1 at services"

    -- Executing [1 at services:1] Answer("SIP/5506-00000014", "500") in new
stack
    -- Executing [1 at services:2] MusicOnHold("SIP/5506-00000014", "") in
new stack
    -- Started music on hold, class 'default', on channel
'SIP/5506-00000014'
    -- Remote UNIX connection disconnected
Got  RTP packet from    192.168.5.134:2224 (type 00, seq 034120, ts
3799238534, len 000160)
Sent RTP packet to      192.168.5.134:2224 (type 00, seq 064889, ts
000160, len 000160)
Got  RTP packet from    192.168.5.134:2224 (type 00, seq 034121, ts
3799238694, len 000160)
Sent RTP packet to      192.168.5.134:2224 (type 00, seq 064890, ts
000320, len 000160)
....etc

Everything is fine and dandy.  Here's the problem:

26213 is a polycom phone on demo1 (which is also asterisk).. and this bug
exhibits itself whether using sip or iax.

on pbx A
asterisk -rx "originate SIP/demo1-sip/26213 extension 1 at services"

    -- Executing [1 at services:1] Answer("SIP/demo1-sip-00000015", "500") in
new stack
    -- Executing [1 at services:2] MusicOnHold("SIP/demo1-sip-00000015", "")
in new stack
    -- Started music on hold, class 'default', on channel
'SIP/demo1-sip-00000015'

no RTP!

Now we try a Playback() beforehand.

on pbx A
asterisk -rx "originate SIP/demo1-sip/26213 extension 2 at services"

    -- Executing [2 at services:1] Answer("SIP/demo1-sip-00000016", "500") in
new stack
    -- Executing [2 at services:2] Playback("SIP/demo1-sip-00000016",
"lunch") in new stack
    -- Remote UNIX connection disconnected
Sent RTP packet to      192.168.15.20:16730 (type 00, seq 049074, ts
000160, len 000160)
    -- <SIP/demo1-sip-00000016> Playing 'lunch.ulaw' (language 'en')
Sent RTP packet to      192.168.15.20:16730 (type 00, seq 049075, ts
000320, len 000160)
Sent RTP packet to      192.168.15.20:16730 (type 00, seq 049076, ts
000480, len 000160)
Sent RTP packet to      192.168.15.20:16730 (type 00, seq 049077, ts
000640, len 000160)
Sent RTP packet to      192.168.15.20:16730 (type 00, seq 049078, ts
000800, len 000160)
....etc
    -- Executing [2 at services:3] MusicOnHold("SIP/demo1-sip-00000016", "")
in new stack
    -- Started music on hold, class 'default', on channel
'SIP/demo1-sip-00000016'
Got  RTP packet from    192.168.15.20:16730 (type 00, seq 017805, ts
2133860346, len 000160)
Got  RTP packet from    192.168.15.20:16730 (type 00, seq 017806, ts
2133860506, len 000160)
Got  RTP packet from    192.168.15.20:16730 (type 00, seq 017807, ts
2133860666, len 000160)
Sent RTP packet to      192.168.15.20:16730 (type 00, seq 049106, ts
005280, len 000160)
....etc

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---------------------------------------------------------------------- 
 (0122828) pabelanger (manager) - 2010-06-02 13:56
 https://issues.asterisk.org/view.php?id=16627#c122828 
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Is this still a problem with 1.6.2? I did not see the same results last
week when testing something MOH related.
--
Per the Asterisk maintenance timeline page at
http://www.asterisk.org/asterisk-versions maintenance (bug) support for the
1.6.0 and 1.6.1 branches has ended. For continued maintenance support
please move to the 1.6.2 branch.

More information on this change can be found in the release announcement:
http://www.asterisk.org/node/49924
 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-06-02 13:56 pabelanger     Note Added: 0122828                          
2010-06-02 13:56 pabelanger     Status                   acknowledged =>
feedback
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