Kevin, thanks for your response.
Here is the more detailed Wireshark capture of the SIP traffic between phone
(10.0.1.57) and Asterisk (10.0.1.103). The numbers between parentheses are
Request Frames. I don't see an ACK for the 200 OK to the INVITE (491) for the
dialplan that gives us Retransm
The timeout value is milliseconds, not seconds. I know that wasn't properly
documented in older versions of Asterisk, but it is at least in 10.1
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman
Sent: Sunday, March 18,
I think I have found a bug in the park command
[Syntax]
Park([timeout][,return_context[,return_exten[,return_priority[,options)
exten => doParkAttempt,n,Park(60,DoMyParkReturn,2003,1,s) Each time I call
it with a timeout value. it fails to use the time out value it is set to 0
and returns
Hello everyone,
I see that the "yum install freepbx" from Digium repository actually
installs the latest FreePBX which is nice. However, I don't see the old FOP
in FreePBX anymore. Is there a way to install FOP or FOP2 through
repository?
Thanks,
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I have a site that moved to the latest 1.8 revision, and began to
have problems with phones in "far away places" (South America,
and the MidEast).
What I see is that when a Dial() is issued, the sip channel driver
sends out an INVITE to the phone. Very soon thereafter, Asterisk
retransmits the
I'm setting up res_fax to use with an iax provider. I'm calling over
PSTN to the provider. When I stand at our fax machine (Brother), I can
see the call come in, and it appears to set up correctly. What is odd,
however, is that asterisk drops off while the fax machine is still
sending. I've low
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