[Asterisk-Users] Hickup and missing voice

[tmpm]

Sorry Rich, both ends use analog bell 500 sets. The system has trunks to 
Iaxtel and FWD as well, but this occurs on all lines, no matter who how or 
where the calls are placed. * to PSTN, PSTN to *, * to *, all the same 
prob. Will look into that CVS. Thanks.



>Several thoughts come to mind, but since you didn't include any comments
>as to what type of phones you're using, etc, these are simply guesses.
>
>1. As someone else mentioned, if you're using an older version of code,
>you might first try upgrading.
>
>2. If you're using Cisco phones, upgrade to latest dev (head) cvs. There
>has been issues with iax/gsm transcoding to sip/rtp that have been slowly
>getting fixed. (Part of those problems relate to how cisco deals with
>uneven timestamps within the rtp stream.)
>
>3. Several phones provide an option that essentially means "transmit silence".
>The default for the option is frequently "transmit silence = no" which will
>cause choppy sound. (In Xlite for exampe, its under Advanced Settings.)
>Change the option to yes.
>
>4. If you have other hubs/switches/infrastructure components involved in
>the end-to-end path, might consider checking those to ensure packets are
>being dropped, collisions are reasonable, etc. One way to check is to use
>ethereal (or some other packet sniffer) to observe a real conversation,
>and look at the flow of packets to ensure they are received/transmitted
>in an even and consistent stream.
>
>Rich
>
>
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