Re: [Freeswitch-users] DTMF Digits Lost when Under Load

2009-11-30 Thread Michael Toop
)   --
ResponseTimeRepartition value=10, 20, 30, 40, 50, 100, 150, 200/

!-- definition of the call length repartition table (unit is
  ms) --
CallLengthRepartition value=10, 50, 100, 500, 1000, 5000, 1/

  /scenario

Cheers,

Michael

On Mon, Nov 16, 2009 at 5:25 PM, Anthony Minessale 
anthony.miness...@gmail.com wrote:

 That's a pretty small problem description to be so sure about something.
 It would probably be better to capture some evidence of the exact problem
 you are having since we are using computers and we need to see the computers
 in action doing something specifically incorrect to diagnose any sort of
 problem.  Take the time to describe the origin and destination of your
 calls, the call flow, the hardware in use on both ends of the call, detailed
 console logs on debug level, (maybe even uncomment the 2833 debug ifded in
 switch_rtp.c) and gather something to go on besides I seem to be losing
 dtmf) maybe a packect capture of the networking interface on both ends of
 these calls.

 Also problems should be reported to http://jira.freeswitch.org not this
 mailing list.
 Save us a step if you report a jira and provide all the info above or we
 will just have to ask for it again.


 On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop 
 micha...@voxcore.voxtelecom.co.za wrote:

 Hi All,

  I have an issue that when my call volumes on my FS IVR box  30 calls
 DTMF digits are lost (using RFC2833).  It is definitely load related as it
 all works perfectly under 30 calls.

  Any pointers or a solution to the problem?

 Thanks,

 Michael
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[Freeswitch-users] B Leg Account Code on Fail Over dialing

2009-11-26 Thread Michael Toop
Hi Everyone,

 How do I get the outbound sofia SIP route that the call took into a CDR
when using fail over dialing with the 'bridge' application?

 ...I have tried numerous approaches with no luck, this last attempt pasted
below did not work either:

dialString =
{provider=providerB}sofia/gateway/sipB/%s|{provider=providerC}sofia/gateway/sipC/%s
% (numberToDial,  numberToDial)
session.execute(bridge,dialString)

 I am using mod_python and this line is in the Python module called in the
dialplan.

Thanks!

Michael
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[Freeswitch-users] DTMF Digits Lost when Under Load

2009-11-16 Thread Michael Toop
Hi All,

 I have an issue that when my call volumes on my FS IVR box  30 calls DTMF
digits are lost (using RFC2833).  It is definitely load related as it all
works perfectly under 30 calls.

 Any pointers or a solution to the problem?

Thanks,

Michael
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Re: [Freeswitch-users] DTMF Digits Lost when Under Load

2009-11-16 Thread Michael Toop
Hi Anthony,

 Thanks for the input.  I will try  reproduce the problem  give you
something more concrete to work with  log it in Jira.

Thanks again,

Michael

On Mon, Nov 16, 2009 at 5:25 PM, Anthony Minessale 
anthony.miness...@gmail.com wrote:

 That's a pretty small problem description to be so sure about something.
 It would probably be better to capture some evidence of the exact problem
 you are having since we are using computers and we need to see the computers
 in action doing something specifically incorrect to diagnose any sort of
 problem.  Take the time to describe the origin and destination of your
 calls, the call flow, the hardware in use on both ends of the call, detailed
 console logs on debug level, (maybe even uncomment the 2833 debug ifded in
 switch_rtp.c) and gather something to go on besides I seem to be losing
 dtmf) maybe a packect capture of the networking interface on both ends of
 these calls.

 Also problems should be reported to http://jira.freeswitch.org not this
 mailing list.
 Save us a step if you report a jira and provide all the info above or we
 will just have to ask for it again.


 On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop 
 micha...@voxcore.voxtelecom.co.za wrote:

 Hi All,

  I have an issue that when my call volumes on my FS IVR box  30 calls
 DTMF digits are lost (using RFC2833).  It is definitely load related as it
 all works perfectly under 30 calls.

  Any pointers or a solution to the problem?

 Thanks,

 Michael
 ___
 FreeSWITCH-users mailing list
 FreeSWITCH-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org




 --
 Anthony Minessale II

 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 Twitter: http://twitter.com/FreeSWITCH_wire

 AIM: anthm
 MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
 IRC: irc.freenode.net #freeswitch

 FreeSWITCH Developer Conference
 sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
 iax:gu...@conference.freeswitch.org/888
 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
 pstn:213-799-1400

 ___
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