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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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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:                     new
Asterisk Version:           SVN 
JIRA:                        
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:              2010-01-17 14:22 CST
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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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---------------------------------------------------------------------- 
 (0116793) kobaz (reporter) - 2010-01-17 14:22
 https://issues.asterisk.org/view.php?id=16627#c116793 
---------------------------------------------------------------------- 
Uploaded sip debug of the problematic call 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-01-17 14:22 kobaz          Note Added: 0116793                          
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