Hi
My call flow is
T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN
Call is placed in reverse direction - from PSTN to T38 Gateway.
T38 GW reinvites with T38, and asterisk passes it along to the SIP provider.
The SIP provider challenges it and asterisk reponds to the Challenge with
INVITE with Auth credentials...however, the Asterisk changes the SDP and
replaces the T38 info in SDP with G711uLaw....and fax fails. How do I configure
the host entry in users.conf such that it maintains the T38 reinvite as it
responds to the SIP INVITE challenge from the Sip Provider.
Her eis my users.conf entry for Asterisk registration to the Sip Provider. (I
know I don't have T38 as allowed codecs, not sure what to add for T38)
[trunk_66]
;register
allow = ulaw
dialformat = ${EXTEN:1}
canreinvite = no
hasexten = no
hasiax = no
hassip = yes
host = provider.com
insecure = very
port = 5060
registeriax = no
registersip = yes
trunkname = abc
username = abc
disallow = gsm,g726,alaw
contact = abc
secret = abc
Any ideas appreciated.
Thx
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