Hello,
any idea why the Asterisk CLI gets flooded by these messages ? How can
the SIP peer /vita3/ cause this flood ?
[Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] --
SIP/vita3-10af requested media update control 26, passing it to
SIP/708708-10b3
[Apr 2 11:45:48]
On Tuesday 02 April 2013, Jonas Kellens wrote:
Hello,
any idea why the Asterisk CLI gets flooded by these messages ? How can
the SIP peer /vita3/ cause this flood ?
First question: What is vita3 ? A hardware SIP phone, a softphone, an ATA
or something else?
--
AJS
Answers come *after*
The SIP peer vita3 is a realtime sip peer, installed in a hardware
IP-phone (Siemens Gigaset N510 pro).
Jonas.
On 04/02/2013 12:35 PM, A J Stiles wrote:
On Tuesday 02 April 2013, Jonas Kellens wrote:
Hello,
any idea why the Asterisk CLI gets flooded by these messages ? How can
the SIP peer
(Message re-ordered for readability. The beginning is *not* the right place
for your response -- answers come *after* questions, or *between* points.)
On Tuesday 02 April 2013, Jonas Kellens wrote:
On 04/02/2013 12:35 PM, A J Stiles wrote:
On Tuesday 02 April 2013, Jonas Kellens wrote:
On 04/02/2013 12:50 PM, A J Stiles wrote:
(Message re-ordered for readability. The beginning is *not* the right place
for your response -- answers come *after* questions, or *between* points.)
On Tuesday 02 April 2013, Jonas Kellens wrote:
On 04/02/2013 12:35 PM, A J Stiles wrote:
On Tuesday
On 04/02/2013 06:37 AM, Jonas Kellens wrote:
On 04/02/2013 12:50 PM, A J Stiles wrote:
(Message re-ordered for readability. The beginning is *not* the right place
for your response -- answers come *after* questions, or *between* points.)
On Tuesday 02 April 2013, Jonas Kellens wrote:
On
01.04.2013 23:52, Paul Belanger пишет:
On 13-04-01 03:16 PM, Dmitriy Serov wrote:
31.03.2013 23:15, Barry Flanagan ?:
On 31 March 2013 18:11, Dmitriy Serov serov@gmail.com
mailto:serov@gmail.com wrote:
Hi, asterisk admin and users.
I need to SIP INVITE uri with domain via peer.
Hi, I'm curious what chip Digium is using in the latest TDM400 cards.
Specifically, to my recollection, they used to use the TigerJet 320G,
however somewhat recently, Tigerjet was bought out, and now the 320G is no
longer produced.
Maybe a better question is: is there a way I can take the latest
I am currently running two different versions of Asterisk
11.0.1
11.2.1
I have noticed the bug occur on both servers.
The issue is that when I try to dial a phone number sometimes the call will
never go out. I will check the Asterisk server with NGREP and see that the
SIP messages are making
