Zeeshan Zakaria wrote:
This is not elastix or FreePBX forum and asking non-asterisk related
questions here is misusing this mailing list.
Since the poster may not be sure this isn't an Asterisk problem, and
Asterisk IS involved, your position is unreasonable.
Self appointed list police aren't received well, and waste archive space.
If one thinks the post isn't appropriate for the list, simply delete and
move on.
JN
Allow anonymous sip is not an asterisk feature. Look in the code in
extensions.conf what it is programmed to do and you'll figure out why
it is happening. Or maybe post the code and ask why such a behaviour,
which'll be better way to ask this elastix related question here. If
you know what this part of dialplan does, rest is easy to figure out.
Zeeshan A Zakaria
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www.ilovetovoip.com <http://www.ilovetovoip.com>
On 2010-09-10 11:17 PM, "bruce bruce" <[email protected]
<mailto:[email protected]>> wrote:
Hi Everyone,
I have a provider whose DID used to come into the box just fine but
recently stopped working. Nothing has been changed on our end.
Here is what I get when doing "sip set debug peer PROVIDER":
Sending to 123.123.123.123 : 5060 (no NAT)
^^^^ That is ALL I am getting with sip debug turned on.
With Allow Anonymous SIP set to YES, then the call comes in properly
and you see the ACK, REQUEST and ACCEPT of sip debug just fine.
This is Elastix with Asterisk 1.4.33.1
Any thoughts?
Thanks
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