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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
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
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 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
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
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
