Thanks for the response. I can not understand why ptime is not an issue for 
voicemail. Also, how come after making the recommended changes to "codec-ms" in 
sofia.conf.xml.vm and internal.xml file, sipx negotiating for ptime 20? Also, 
is 
there something that I need to do after making these changes? For example, 
system prompted me to do "freeswitch.sh --configtest"...And like I said, I 
rebooted the box a few times as well.

I also saw that reload/rescan a command "sofia profile <profilename> 
reload.xml". However, there is no command called sofia in the sipxbox. Is there 
something I need to compile/download before I can run this command?

Please advise,

Thanks and have a nice day,

-SrinivasaRao Seelam



________________________________
From: Joegen Baclor <[email protected]>
To: srinivasa rao <[email protected]>
Cc: Discussion list for users of sipXecs software 
<[email protected]>; [email protected]; Raju Sunkasari 
<[email protected]>; Ram N <[email protected]>; Mala Sistla 
<[email protected]>
Sent: Thu, January 27, 2011 11:41:07 PM
Subject: Re: [sipx-users] auto-attendant messages are played real fast

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 <[email protected]>
>To: Discussion list for users of sipXecs software 
><[email protected]>
>Cc: srinivasa rao <[email protected]>
>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. 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
_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/

Reply via email to