According to https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
Asterisk 10 was EOLd on 2013-12-15 and has been on security fix only for a
year before that. If you find the bug and figure out how to fix it, the fix
will never be released because Asterisk 10 is EOLd.Take a look
Hi Daniel
Le 14/02/2014 07:33, Daniel van den Berg a écrit :
Hi All,
Lets say I want to setup a queue that will handle inbound calls to
dynamically added agents that are all mobile numbers. Now when I do this
setup it works, it loads the agents dynamically and if the mobile phone
is on and
On 14/02/14 06:33, Daniel van den Berg wrote:
Hi All,
Lets say I want to setup a queue that will handle inbound calls to
dynamically added agents that are all mobile numbers. Now when I do this
setup it works, it loads the agents dynamically and if the mobile phone
is on and have reception it
On 14/2/14 9:21 am, Gareth Blades wrote:
I would suggest using the 'M' option on the Dial command to run a macro.
The macro can just wait fir a key to be pressed and until it is pressed
the Dial is still effectively ringing. So if it does go to voicemail
then the call wont get put through. You
Hi all,
How does one detect the 'divert' to voicemail?
Say we have PRI lines and as wel as SIP Trunks to connect to mobile phones.
How can asterisk know if the call is being diverted??
On 14 February 2014 10:11, Chris Bagnall aster...@lists.minotaur.cc wrote:
On 14/2/14 9:21 am, Gareth
On 14/2/14 10:54 am, Tiago Geada wrote:
How does one detect the 'divert' to voicemail?
If you're using the mobile network's voicemail service, you can't as a
general rule; you've no reliable way of knowing whether that call was
answered by the user or their voicemail service.
However, if
On 13 Feb 2014, at 09:55, Aldo Bergamini aabe...@gmail.com wrote:
Hi,
I did compile the latest DAHDI and LibPRI, with no success… So I thought
about updating the Asterisk package to the last known 1.6.2 release.
Now it's crashing at some different point.
This is the the strace
- Original Message -
SIP options message is due to check the peer registration is
keepalive. As per my understanding it might be because of network
flap may be wireshark trace can give you any clue.
Regards
Correct. I understand the role and function of the OPTIONS requests. The
On Friday 14 Feb 2014, Tiago Geada wrote:
Hi all,
How does one detect the 'divert' to voicemail?
Say we have PRI lines and as wel as SIP Trunks to connect to mobile phones.
How can asterisk know if the call is being diverted??
It can't.
But you know (from the STD code) whether the
On 13 Feb 2014, at 18:10, Tim Nelson tnel...@rockbochs.com wrote:
I recently experienced an odd situation. I have an Asterisk 11.5.0 system
(Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At
some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box
On Friday 14 Feb 2014, Tiago Geada wrote:
Hi all,
How does one detect the 'divert' to voicemail?
Say we have PRI lines and as wel as SIP Trunks to connect to mobile
phones.
How can asterisk know if the call is being diverted??
It can't.
But you know (from the STD code)
I believe I am running an AGI (to put users in a conf) before the
confbridge is built. So the users are not really get in the conf...
exten X,1,run agi to put users in conf
exten X,n,ConfBridge()
How do I have in the dial plan ConfBridge() and someplace
run an AGI that brings the users I want
On Friday 14 Feb 2014, Jerry Geis wrote:
I believe I am running an AGI (to put users in a conf) before the
confbridge is built. So the users are not really get in the conf...
exten X,1,run agi to put users in conf
exten X,n,ConfBridge()
How do I have in the dial plan ConfBridge() and
I have a customer with a more or less unique need. Right now we
are using Wombat as a dialer software so they can contact clients for QA
purposes. Everything is working very well and their contact center
productivity is way up from the old manual dialing method.
The only thing we
Have a look at vicidial it has alternate number dialing capability.
Mituo
On Saturday, February 15, 2014, Carlos Chavez cur...@telecomabmex.com
wrote:
I have a customer with a more or less unique need. Right now we are
using Wombat as a dialer software so they can contact clients for QA
Hello,
I inherited an Asterix phone system. I am well versed in Windows based
platforms but have zero experience in Linux and Asterix, no make matters worse
I have no documentation on this system. I had to change the entire networks
gateway address for various reasons but now the Asterix
On Fri, Feb 14, 2014 at 5:40 PM, Dave Swangler ctit...@live.com wrote:
Hello,
I inherited an Asterix phone system. I am well versed in Windows based
platforms but have zero experience in Linux and Asterix, no make matters
worse I have no documentation on this system. I had to change the
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