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Summary:ASTERISK-09646: StartMusicOnHold application stops inband ringing on all extension
Reporter:Federico Santulli (fsantulli)Labels:
Date Opened:2007-06-11 11:04:11Date Closed:2011-06-07 14:02:48
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I'm dialing a sip extension in the dialplan and i've enabled earlyrtp in sip.conf so, when i ask for StartMusicOnHold everywhere in the dialplan, magically i didn't hear the ring anymore.

****** ADDITIONAL INFORMATION ******

Tested on Asterisk 1.2.18
Comments:By: Joshua C. Colp (jcolp) 2007-06-11 11:22:54

We need much more information. sip.conf, console output, call flow.
By: Federico Santulli (fsantulli) 2007-06-11 12:33:57

This is extensions.conf

[default]
include => incoming
include => internals

[incoming]
exten => s,1,Answer
exten => s,2,Dial(SIP/int11,120,m(test)) ; It seems to be due by dial app. error
exten => s,3,Wait(15)
exten => s,4,Hangup
exten => s,103,Busy

[internals]
exten => 11,1,Dial(SIP/int11,120)
exten => 11,2,Hangup
exten => 11,102,Busy

exten => 12,1,Dial(SIP/int12,120)
exten => 12,2,Hangup
exten => 12,102,Busy


By: Federico Santulli (fsantulli) 2007-06-11 12:39:33

this is sip.conf

[general]
context=incoming
bindport=5060
bindaddr=0.0.0.0
maxexpiry=3600
minexpiry=60
disallow=all
allow=alaw
allow=ulaw
language=it
rtptimeout=60
rtpholdtimeout=360
progressinband=no     ; As we need to hear telco ringings or unreachable messages

[int11]
type=friend
context=internals
callerid=Internal 11 <11>
dtmfmode=rfc2833
regexten=11
username=int11
secret=[hidden]
insecure=invite,port
host=dynamic
callgroup=1
pickupgroup=1
canreinvite=no

[int12]
type=friend
context=internals
callerid=Internal 12 <12>
dtmfmode=rfc2833
regexten=12
username=int12
secret=[hidden]
insecure=invite,port
host=dynamic
callgroup=1
pickupgroup=1
canreinvite=no


By: Federico Santulli (fsantulli) 2007-06-11 12:47:10

this is asterisk.conf

[options]
internal_timing = yes (as we're using sip channels for incoming and we need a clear moh)


By: Federico Santulli (fsantulli) 2007-06-11 13:17:10

this is callflow !!!

***********
HERE SIP/int11 is dialing SIP/int12
***********

   -- Executing Dial("SIP/int11-b6c05510", "SIP/int12|90") in new stack
   -- Called int12
   -- SIP/int12-08235a20 is ringing

***********
THIS IS AN INCOMING CALL FROM SIP/TESTPEER
***********

   -- Executing Answer("SIP/testpeer-b6c39a58", "") in new stack
   -- Executing Dial("SIP/testpeer-b6c39a58", "SIP/int11|120|m(test)") in new stack
   -- Called int11
   -- Started music on hold, class 'test', on channel 'SIP/testpeer-b6c39a58'
   -- Got SIP response 486 "Busy here" back from 10.10.10.169
   -- SIP/int11-082295f0 is busy
 == Everyone is busy/congested at this time (1:1/0/0)
   -- Stopped music on hold on SIP/testpeer-b6c39a58
   -- Executing Wait("SIP/testpeer-b6c39a58", "15") in new stack
 == Spawn extension (internals, 12, 1) exited non-zero on 'SIP/int11-b6c05510'
 == Spawn extension (incoming, s, 3) exited non-zero on 'SIP/testpeer-b6c39a58'
By: Federico Santulli (fsantulli) 2007-06-11 13:20:10

It seems that the asterisk inband generated ringing to SIP/int11 stops when

Dial(SIP/int11,120,m(test))

is executed.

In musiconhold.conf class test is defined and works.

Take note i'm using the internal_timing patch available at http://bugs.digium.com/view.php?id=5374


By: Joshua C. Colp (jcolp) 2007-06-12 10:23:27

Asterisk is not generating the ring, the phone is. It could be possible that when it sends back the Busy it screws up it's own ring generation... hrm. Can you please ATTACH a sip debug as well?
By: Joshua C. Colp (jcolp) 2007-06-22 11:33:31

It's been 10 days now without any update positive or negative. If you still believe this is an Asterisk issue and have more information, please reopen.