Hi List,
I have a Asterisk + FreePbx Server setup with around 10 SIP extensions
and 1 VoIP trunk (CordiaVoIP), when we dial-out to any number call is
being dropped with the following message on asterisk log:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called CordiaVoIP
Your trunk shows busy:
* -- Called CordiaVoIP/639285010430
-- SIP/CordiaVoIP-0015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)*
Try this in the CLI (asterisk -r):
*core set verbose 0*
*sip set debug peer CordiaVoIP*
And then make a call and read why t
Sorry I do not understand it, here is result after:
Audio is at 172.16.9.15 port 15022
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.211.94.211:5060:
INVITE sip:639285010...@lasip1.cordiaip.net SIP/
Bruce,
Thanks. I already figured out the problem. It seems that a firewall issue.
Regards,
Malvin
On 7/13/2011 12:30 PM, Bruce B wrote:
Your trunk shows busy:
*/ -- Called CordiaVoIP/639285010430
-- SIP/CordiaVoIP-0015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1
Yes, that is it. And you were inviting the provider to contact you back at
your private subnet of 172.16.x.x:
*From: "Cordia" ;tag=**as2267fdcc*
*
*
So, hence their responces never made it back to you and that's why you are
re-transmitting 6 times to get attention.
*
*
- Bruce
On Wed, Jul 13, 201
