I think the reINVITe came from sipXbridge. Not so sure if this is
something hard-coded but perhaps it's the reason why ptime is set to 20
in freeswitch config as the default in the first place. Bottom line,
this is not really the fault of either sipx or freeswitch. Your
provider is not complying to standards and their switch seems to be a
Sonus. To quote Tony G. Run away!
On Friday, 28 January, 2011 11:16 AM, srinivasa rao wrote:
Problem is still there even after changing the "codec-ms" value to 30..
1) I changed the "param name="codec-ms" value="30"/".....And rebooted
the system. System prompted me to run "freeswitch.sh --configtest".
And the issue */_(autoattendant messages are played at very fast
rate)_/* there.
2) Afterthis, I called bandwidth again; bandwidth asked me to change a
value of <param name="inbound-codec-negotiation"
value="*/_greedy_/*"/> in the
freeswitch/conf/sip_profiles/internal.xml file. Therefore, I rebooted
the server. Also, in the management interface, I clicked on the server
and click on send profiles as well. This did not fix the problem either.
3) The interesting thing is that I went to dial profiles and disabled
auto attendant and only enabled voice mail. And the "voicemail
message" is played very very very clear. Then I called bandwidth to
run another trace to see what is going on with "ptime" for voicemail.
Surprisingly, bandwidth told me that ptime negotiation is not a
problem for the voicemail. I have attached both call traces provided
by bandwidth.com.
Could you please look at these trace and kindly advise what I need to
do for the autoattendant issue?
Thanks and have a nice day,
-SrinivasaRao Seelam
------------------------------------------------------------------------
*From:* Joegen Baclor <jbac...@ezuce.com>
*To:* Discussion list for users of sipXecs software
<sipx-users@list.sipfoundry.org>
*Cc:* srinivasa rao <ssv...@yahoo.com>
*Sent:* Wed, January 19, 2011 6:51:43 PM
*Subject:* Re: [sipx-users] auto-attendant messages are played real fast
On Thursday, 20 January, 2011 12:26 AM, srinivasa rao wrote:
Could anyone please help me with changing the ptime value in sipx?
We have our VoIP router from bandwidth.com <http://bandwidth.com/>.
They changed their ptime setting from 20 to 30 in the mid of december
2010.
Bandwidth also told me the following:
1) Call comes in; Bandwidth requests for ptime of 30 and sipx accepts it.
2) Then sipx sends an invite with a ptime of 20.
Please advise,
Thanks and have a nice day,
SrinivasaRao Seelam
Look for sofia.conf.xml.vm file in the sipx config folder. Add the
folowing parameter right after the codec setting
<param name="codec-prefs"
value="$settings.getSetting('FREESWITCH_CODECS').Value"/>
<!-- This forces ptime to 30 -->
<param name="codec-ms" value="30"/>
RESTART
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