Hi,
I'm using Cicso - Linksys SPA3102 to connect to asterisk. I have followed
the official user manual and the blog post here
http://www.skelleton.net/2012/08/02/linksys-spa-3102/
When I call an extension say 225 from the analog phone, I can get the IVR I
have setup in my dialplan. But when I
Face wrote:
Well, thanks for responding. I went back to 10.10.0 and things seem to
be working fine now!
This is certainly good to know but I'd like to know why upgrading to 11
did not seem to work for you. This is the first case since it's been out
where it doesn't appear to have been
I upgrading to 11 because I want to use the MessageSend command from the
AMI, ver 10 dose not have MessageSend In the list of
commands. Unfortunately I remove ver 11 and I dont think I can provide the
information you asked.
On Tue, Nov 20, 2012 at 4:46 PM, Joshua Colp jc...@digium.com wrote:
Face wrote:
Hello,
Hola,
After Upgrade to Asterisk 11.1.0-rc1 I keep getting
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [603@DLPN_AlDimnaDialPlan:601]
Dial(SIP/601-0002, SIP/603)
On Mon, Nov 19, 2012 at 3:51 PM, Joshua Colp jc...@digium.com wrote:
Face wrote:
Hello,
Hola,
After Upgrade to Asterisk 11.1.0-rc1 I keep getting
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Hello,
After Upgrade to Asterisk 11.1.0-rc1 I keep getting
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [603@DLPN_AlDimnaDialPlan:601]
Dial(SIP/601-0002, SIP/603) in new stack
[Nov 16
