asterisk users - Apr 2004 - Timing

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Hi,

As I understand it, Asterisk currently uses the timestamps in incoming RTP 
packets to build outgoing voice frames. Is this true?

Would it be possible for me to use i.e. zaprtc as a timing source for the 
outgoing stream? I.e. in a setup like below I'd like to use zaprtc timing on
Ast1 because I don't trust the timestamps coming from the SIP client.

SIP client --- Ast1 --- IAX2 --- Ast2 --- Zap --- PRI

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374

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I?m trying to get busydetect working with an analog PBX. I have an 
extension connected to a X100P. PBX generates a 230ms tone, followed by 
70ms silence, another 230ms tone, 70ms silence and keeps repeating the 
pattern.
I?m not sure if I understand how BUSY MAX, MIN and THRESHOLD  in dsp.c 
works, because I have the default values from 0.9.0 and it does not 
detect the busy signal:

#define BUSY_PERCENT            10
#define BUSY_THRESHOLD          100
#define BUSY_MIN                75
#define BUSY_MAX                1100

Can someone with more experience intunnig these parameters suggest me 
the right values I should use?

Thanks
Gelson

Second try; I got no answer on this one:

  I?m trying to get busydetect working with an analog PBX. I have an
  extension connected to a X100P. PBX generates a 230ms tone, followed by
  70ms silence, another 230ms tone, 70ms silence and keeps repeating the
  pattern.
  I?m not sure if I understand how BUSY MAX, MIN and THRESHOLD  in dsp.c
  works, because I have the default values from 0.9.0 and it does not
  detect the busy signal:

  #define BUSY_PERCENT            10
  #define BUSY_THRESHOLD          100
  #define BUSY_MIN                75
  #define BUSY_MAX                1100

  My understanding is that the tone I get is inside the max/min timing, 
   isn?t it?

  Can someone with more experience in tunnig these parameters suggest me
  the right values I should use?

  Thanks
  Gelson

I?m using * 0.9.0 and have a X100P connected to my analog PBX. I can?t 
detect hangups on this line, so I turned on busydetect=yes in 
zapata.conf. I also have busycount=6.
	While the line is connected to the PBX, I can never detect busy and the 
line hangs at the end of every call. If I connect the same X100P to the 
telco line, without the PBX, then it can detect busy and hangs up the 
line after 6 busy tones, as expected.
	I have recorded the busy tones and found that telco uses a standar one 
(250ms tone, 250 ms silence). My PBX, however, is using a 120ms tone, 
80ms silence sequence.
	How can I adjust the detection routines to the tone I?m getting?  I 
have tried to mess with busy_min, busy_max etc on dsp.c with no luck. 
I?m sure I doesnt really understand the meaning of those parameters.
	A also tryed to compile using TONE_ONLY  but it gives a compilations error.
	Can someone suggest what times should I been using?

	Thanks a lot.

	Gelson