[Asterisk-Dev] SIP DTMF causing voice quality loss?

[Steve Radich]

I've been playing with SIP a lot lately and we rolled out some more of our
employees on SIP based long distance (testing to see if it's feasible to
collocate our Asterisk at our data center with a PRI - hence the talks
before about sharing PRI, etc.)..

 

I'm seeing an odd issue, I'm testing through IConnectHere with limited
success - however I seem to have one thing that may be reproducible (too
early to say if reproducible):

            Layout:  POTS based phone, hooked to Zhone Channel bank - Zhone
to Quad T1 Digium Card on Asterisk server.  Asterisk then Dial/SIP to analog
based phone system.  

Trying: Analog phone system (called party) says press 123 for sales.. We
press 123.  Suddenly call quality is horrible (we can hear fine about 75% of
the time, other 25% no audio, called party says can't hear us hardly) -
other DTMF tones don't seem to go through right (maybe unclear so not being
recognized).

Works: Calling what I would guess is another digital PBX works fine using
this same layout.  I've tried a few banks / large businesses and no
problems.

Works: Can call a phone via SIP and recognize DTMF from called party (i.e. #
to transfer)

 

Any ideas?


We should have another SIP based gateway up next week and we now have
pricing on DC based PRI that's good so we may just go with PRI.  I'd rather
avoid the cross connect fees / loops on the PRI since we can get SIP based
easier and have plenty of bandwidth - it seems like just another thing to go
wrong in Asterisk using PRI instead of Cisco delivered SIP (maybe I'm
backwards on this).

 

PS: How does IAX handle DTMF between Asterisk servers?

Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers 
BitShop, Inc. -  <http://www.bitshop.com/> http://www.bitshop.com -
$149/month colo special

 

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