Not using the CDR for billing, but I do use it to see usage and to
know if it's cheaper to purchase a provider with unlimited incoming
and pay-per-minute outgoing. I disabled 'SIP Transformation' in the
SonicWall and so far so good (10/10 calls worked, more testing to be
had, stay tuned.)
On Sat,
Usually, calls terminating at 30 seconds is a sure sign that you need to add
an Answer() in your dialplan. Try dropping that in before you dial out. I
have seen this so many times and Answer() has always fixed the issue. The
magic number is 30 seconds.
Depending on if you use your CDRs for
Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode)
with a public IP address. We have our phone system setup as 172.16.2.x
that connect through the SonicWall to Asterisk. Incoming calls work
flawlessly and we no longer get one-way audio. We are only using SIP
(3 trunks now,
Kurt Knudsen wrote:
Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode)
with a public IP address. We have our phone system setup as 172.16.2.x
that connect through the SonicWall to Asterisk. Incoming calls work
flawlessly and we no longer get one-way audio. We are only using
At 14:15 11/7/2008, SIP wrote:
Kurt Knudsen wrote:
Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode)
with a public IP address. We have our phone system setup as 172.16.2.x
that connect through the SonicWall to Asterisk. Incoming calls work
flawlessly and we no longer
Ok, recompiling it now with a 1 instead of XMIT_CRITICAL. Will check
back to see if it worked. Would be nice if it did :)
Thanks,
Kurt
On Fri, Nov 7, 2008 at 3:38 PM, Doug [EMAIL PROTECTED] wrote:
At 14:15 11/7/2008, SIP wrote:
Kurt Knudsen wrote:
Specs: Asterisk 1.4.22 running behind a
That seems to have sort of worked. It seems the phone decided to end
the call this time, instead of Asterisk and now the call is dangling
inside of 'sip show channels'.
So that solution didn't work :(
On Fri, Nov 7, 2008 at 4:28 PM, Kurt Knudsen [EMAIL PROTECTED] wrote:
Ok, recompiling it now
To get to the bottom of it I'd recommend determining why the ACKs are
not getting through to Asterisk rather than trying to work around it.
I'm actually suprised Asterisk terminates the call by default when it
doesn't get the ACK to it's 200 Ok response that must be new for
1.4.22 as I haven't
