BTW, you can use Karoo bridge to hide these funky record routes from the
uncompliant phones.
On 06/12/2012 09:33 PM, Joegen Baclor wrote:
Indeed. The route header is blank and this gives it 0% chance of ever
working with sipx. sipXecs record routes are divine. Thou shalt not
mess with it.
On 06/12/2012 02:44 PM, milosz wrote:
It is probably not getting or missing an ack to indicate the call established.
This would indicate a carrier or gateway config issue.
the m3 seems to be blanking the route header field. take a look at the trace.
I am curious what this actually has to do with sipX?
answer: something.
Adtran's SIP stack, while it has gotten MUCH better, is still garbage (when
used as a gateway). It was designed with Asterisk in mind and is pretty much
limited to routing by numbers only (which makes a lot of alpha-numeric stuff
sipX does not work properly).
i haven't had too many issues, but i've only used it for very basic
deployments. it is cheaper and less horrible to configure than the
audiocodes and is cheaper than and has a superior timing source to the
basic patton 4960. the stack on the patton is the best but it's often
difficult to justify spending 4x the money.
the call disconnect behavior is not limited to the adtran, i had it
happen with the patton as well.
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