By default AMP had NAT=yes in sip.conf, I read in some posts to change
it to one, i was just trying my luck if that works. I have tried
NAT=yes, The Phone gets registered, I can also make recieve calls
but as soon as the call is picked I dont hear anything at both ends.
Does this have anything to do with codecs?
ThanksOn 11/22/05, C F [EMAIL PROTECTED] wrote:
On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All,I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution.
My Asterisk is on a public domain, there is no NAT or firewall in front ofIf no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to
recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot
hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2)with [EMAIL PROTECTED] installed.my Sip.conf :
[2008] ;(Sipura2002)username=2008type=friendsecret=2008record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED]
host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid=device 2008[2009] ;X-Lite Soft Phoneusername=2009type=friend
secret=2009record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED]host=dynamicdtmfmode=rfc2833context=from-internal
canreinvite=nocallerid=device 2009Thanks in advance.. ___ --Bandwidth and Colocation sponsored by
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