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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Anyone for this ?
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 05, 2009 11:02 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] T38 REINVITe issue
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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Ujjval Karihaloo wrote:> > 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 >You'll need to put t38pt_udptl = yes somewhere in your sip.conf, probably in the general section for T.38 to work properly. -- Trevor Peirce Digital Conceptions Canada http://www.digitalcon.ca 1-888-606-3030 / 250-391-7822