[asterisk-bugs] [Asterisk 0015975]: Unable to change the packetization settings (ptime) for codecs from default of 20ms

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=15975 
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Reported By:                jehanzeb
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15975
Category:                   Codecs/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           Older 1.4 
JIRA:                        
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-09-28 05:27 CDT
Last Modified:              2009-09-28 09:16 CDT
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Summary:                    Unable to change the packetization settings (ptime)
for codecs from default of 20ms
Description: 
Hi, I am currently running Asterisk version 1.4.21
my problem is that even though i have tried to force outbound calls with a
codec packetization rate of 10ms, or 30ms, asterisk keeps sending the
Invite message with the default ptime of 20ms.

my sip.config file for this peer is 

[sylantro]
type=friend
disallow=all                    ; First disallow all codecs
disallow=gsm
allow=ulaw:10,alaw:30           ; Allow codecs in order of preference
autoframing=yes
context=testcontext
trustrpid = yes
;dtmfmode=inband
canreinvite=yes
host=195.219.133.219
port=5065

sip show peer command shows the following settings

Name       : sylantro
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Context      : testcontext
  Subscr.Cont. : <Not set>
  Language     : en
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : Yes
  PromiscRedir : Yes
  User=Phone   : No
  Video Support: No
  Trust RPID   : Yes
  Send RPID    : Yes
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : auto
  LastMsg      : 0
  ToHost       : 195.219.133.219
  Addr->IP     : 195.219.133.219 Port 5065
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username:
  SIP Options  : (none)
  Codecs       : 0xc (ulaw|alaw)
  Codec Order  : (ulaw:10,alaw:30)
  Auto-Framing:  Yes
  Status       : Unmonitored
  Useragent    :
  Reg. Contact :
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---------------------------------------------------------------------- 
 (0111427) lmadsen (administrator) - 2009-09-28 09:16
 https://issues.asterisk.org/view.php?id=15975#c111427 
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Could you provide the console output as well, in addition to the sip
history of this call? I can't tell which peer is being used in the outgoing
INVITE with this trace. Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-28 09:16 lmadsen        Note Added: 0111427                          
2009-09-28 09:16 lmadsen        Status                   new => feedback     
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