Thanks again Anthony !
You fixed the issue with DTMF i had reported :
http://jira.freeswitch.org/browse/FSCORE-251
Chris Danielson added to Wiki a nice page collecting these issues with
Sonus :
http://wiki.freeswitch.org/wiki/RTP_Issues
Cheers,
El mié, 10-12-2008 a las 03:10 +0100, Angel
Thanks Anthony , you did a great work ! this is fixed in svn r10691.
Some notes for people using Sonus and L3 as was my case :
in var.xml in some scenario you may need :
X-PRE-PROCESS cmd=set data=send_silence_when_idle=400/
in sip_profiles/internal.xml :
param name=rtp-rewrite-timestamps
most likely it's because during the time you are dong artificial ringback
the other side is not doing RTP right.
When the call is answered we flush the rtp buffer and your missing audio is
probably flushed with it.
so you can choose to have a 3 second delay or erase the 3 seconds as it does
now.
Hi guys,
I've a strange issue with FS , version svn -r10584 ,
when FS bridges a call first 3 seconds of audio are missing , looks that
only happens on PSTN calls and using ringback or transfer_ringback. This
only happens in calls from PSTN , not from VOIP. Some scenarios i tried
to isolate
what does PSTN represent?
I know what the PSTN is but how are you reaching it?
is it TDM, SIP etc... what gateway type other details.
On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero [EMAIL PROTECTED] wrote:
Hi guys,
I've a strange issue with FS , version svn -r10584 ,
when FS bridges a
No TDM , all is SIP :
PSTN --- Sip Proxy_A -- FS ( brigde ) ringback/transfer_ringback
- Sip Proxy_B -- PSTN
In logfile i think you can get some details about Media Gateways
( Sonus ) PSTN inbound / outbound is provided by Level3.
I can get a capture of a call if you want, in capture the
