[Asterisk-Users] SIP to IAX - forcing codec pass thru

[Peter Gradwell]

hi

We take calls inbound via SIP from our Cisco PSTN gateways, and pass it 
to customers using IAX (they run their own asterisk servers).

We've noticed that asterisk is transcoding the call into a different 
codec, if the customer prefers a codec different to that which our cisco 
gw prefers. As such, the quality of the call can degrade.

We'd rather asterisk just passed through the RTP stream and maintained 
the same codec, so that all asterisk did was signalling conversion.



sip.conf...

---

[sip-router-1.gradwell.net]
context=sip-inbound
type=peer
host=sip-router-1.gradwell.net

[sip-router-2.gradwell.net]
context=sip-inbound
type=peer
host=sip-router-2.gradwell.net

---

iax.conf...

[general]
bandwidth=high
disallow=lpc10
jitterbuffer=yes
dropcount=2
maxjitterbuffer=500
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1
tos=lowdelay


---

when a call comes in, we dial something like this, in our dial plan:

     -- Executing Goto("SIP/213.166.5.134-118f5310", 
"sip-users|7770002|1") in new stack
     -- Goto (sip-users,7770002,1)
     -- Executing Dial("SIP/213.166.5.134-118f5310", 
"IAX2/user:pass at customeripaddress/441376350002") in new stack
     -- Called user:3l3phant at customeripaddress/441376350002
     -- Call accepted by customerip (format alaw)
     -- Format for call is alaw
     -- IAX2/customerip:4569-23 answered SIP/213.166.5.134-118f5310

thanks
peter

-- 
peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/
  -- engineering & hosting services for email, web and voip --
   -- http://www.peter.me.uk/  -- http://www.voip.org.uk/ --