Hi,
I have recently took up this project to set up asterisk for my company.
Everything seemed to fine initially but now I am having a strange problem. Here is my setup. I have 2 extensions 1000 and 1001. The first call from either of the extensions goes just fine but when I hang up and do a redial the call doesnt go but the phone says calling… . When I check the asterisk CLI for error I get this == Using SIP RTP CoS mark 5.
If I dial again after 15-20 mins the call goes again everything works fine.
I am using Grandstream GXP285. Please help me out. I am not getting the solution in any of the forums.
Here my sip.conf and asterisk.conf.
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Sip.conf
[incoming]
exten => 1000,1,Dial(SIP/1000,20)
exten => 1000,n,VoiceMail(1000@wc-voicemail,u)
exten => 1000,n,Hangup()
exten => 1001,1,Dial(SIP/1001,20)
exten => 1001,n,VoiceMail(1001@wc-voicemail,u)
exten => 1001,n,Hangup()
exten => 6500,1,Answer(500)
exten => 6500,n,VoiceMailMain(@wc-voicemail)
extensions.conf
[1000]
type=friend ; Friends place calls and receive calls
context=incoming ; Context for incoming calls from this user
secret=Wavecrest@123
callerid=1000
nat=yes
host=dynamic ; This peer register with us
insecure=no
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
disallow=all
allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
progressinband=no