[asterisk-users] iax: unable to transfer - one way audio
Sean Darcy
seandarcy2 at gmail.com
Fri Sep 27 20:08:20 CDT 2013
We have zoiper connected over iax to asterisk in Sydney. The call is to
asterisk in New York. The caller in NZ can hear clearly. Nothing in NY.
Here's the sydney server:
-- Accepting AUTHENTICATED call from <zoiperipaddr>:
> requested format = speex,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (silk16|ulaw|gsm|g722),
> priority = mine
-- Executing [8447 at nz-in:1] Dial("IAX2/n4-270", "IAX2/sydney") in
new stack
-- Called IAX2/sydney
-- Call accepted by <nyipaddr> (format ulaw)
-- Format for call is (ulaw)
-- IAX2/sydney-8819 is ringing
-- IAX2/sydney-8819 answered IAX2/n4-270
-- Channel 'IAX2/n4-270' unable to transfer
-- Channel 'IAX2/sydney-8819' unable to transfer
-- Channel 'IAX2/sydney-8819' unable to transfer
-- Channel 'IAX2/sydney-8819' unable to transfer
The NY server:
-- Accepting AUTHENTICATED call from <sydneyipaddr>:
-- > requested format = ulaw,
-- > requested prefs = (ulaw|silk16|gsm|g722),
-- > actual format = ulaw,
-- > host prefs = (ulaw|gsm|g722),
-- > priority = mine
-- Executing [s at incoming-nz:1] Goto("IAX2/home-2152",
"incoming,s,nz-in") in new stack
-- Goto (incoming,s,5)
-- Executing [s at incoming:5] Dial("IAX2/home-2152",
"DAHDI/g0&SIP/250&SIP/251,60,tT") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called DAHDI/g0
-- Called SIP/250
-- Called SIP/251
-- DAHDI/1-1 is ringing
-- SIP/251-0000001d is ringing
-- SIP/250-0000001c is ringing
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 answered IAX2/home-2152
-- Channel 'IAX2/home-2152' unable to transfer
-- Hanging up on 'DAHDI/1-1'
Any help appreciated.
sean
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