== Using SIP RTP CoS mark 5

Asterisk Asterisk Support
1

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.

======
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

[1001]
type=friend ; Friends place calls and receive calls
context=incoming ; Context for incoming calls from this user
secret=Wavecrest@123
callerid=1001
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

2

Sorry, but you will have to explain a bit more about your setup and provide some more debugs.

Are the calls between your SIP phones not working OK or are you having problems with calls to your telephony provider?

Please provide the CLI output when you make the calls.

3

I havent connected my sip phones to the telephone provider yet. As of now it is only internal extensions.

When I dial the extension the status on my grandstream phone shows “calling…” but the other phone doesnot ring. This does not happen always though. The first call goes fine, then I hang up and redial immediately. then the problem shows up. If I dial again after 15-20 min everything will be back to normal.

When I look at the CLI this is what i get and nothing esle.

== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5

4

Did you check the status of your firewall on the Asterisk server? Try disabling it.