On 08/11/2011 02:03 AM, Jim Boykin wrote:
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is
12 aug 2011 kl. 14:51 skrev Kevin P. Fleming:
On 08/11/2011 02:03 AM, Jim Boykin wrote:
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Hi,
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is that whenever call
is received from the
Anyone?
On Thu, Aug 11, 2011 at 12:33 PM, Jim Boykin boykin...@gmail.com wrote:
Hi,
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
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Web
The problem seems like asterisk is not authenticating at all. It
accept the default invite and transfer it to default contact. ANy
help.
On Thu, Aug 11, 2011 at 12:33 PM, Jim Boykin boykin...@gmail.com wrote:
Hi,
We have difficulty setting up the incoming termination for our
clients. Both
for start you could disable guest access in sip.conf, I guess you do not
need it
On 2011.08.11 14:29, Jim Boykin wrote:
The problem seems like asterisk is not authenticating at all. It
accept the default invite and transfer it to default contact. ANy
help.
On Thu, Aug 11, 2011 at 12:33 PM,
