Hello,
I have installed Asterisk 1.6.26. And I have Digium - TE110P card.
When I call to SIP to PSTN or vise versa, the audio comes only from the PSTN side only.
No audio is coming from SIP softphone.
I can hear everything in softphone . But the audio is not going from Softphone.
Even SIP softphone to SIP softphone audio is not working.
Please help me for this issue.
Bellow is my configuration.
sip.conf
[general]
context=default ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
bindport=5060
bindaddr=0.0.0.0
srvlookup=no ; Enable DNS SRV lookups on outbound calls
videosupport=yes ; Turn on support for SIP video
canreinvite=yes ; Asterisk by default tries to redirect the
disallow=all
;allow=all
allow=alaw
allow=ulaw
;allow=g729
;allow=h264
[4000]
type=friend
username=4000
secret=1234
context=default
host=dynamic
canreinvite=yes
disallow=all
;allow=all
allow=alaw
allow=ulaw
;allow=g729
[5000]
type=friend
username=5000
secret=1234
context=default
host=dynamic
canreinvite=yes
disallow=all
;allow=all
allow=alaw
allow=ulaw
;allow=g729
========================
Thanks in advance.
-Urmi