On 04/02/2013 05:42 PM, Matthew Jordan wrote:
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 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 /vita3/ cause this flood ?
First question: What is "vita3" ? A hardware SIP phone, a softphone, an
ATA or something else?
The SIP peer vita3 is a realtime sip peer, installed in a hardware
IP-phone (Siemens Gigaset N510 pro).
Have you any other Siemens Gigaset N510 pro phones in your setup?
Yes there are. But I want to know what these messages on the CLI mean ?
The device communicating with Asterisk over SIP channel
SIP/vita3-000010af had a change in the media source (26 ==
AST_CONTROL_SRCCHANGE). This occurs when the SSRC in an RTP packet sent
by that device changed.
When in the middle of a dialling operation, we tend to log out when one
of the parties passes information to the other party. In general, this
wouldn't flood the CLI, as a party shouldn't be passing much information
off to the other parties involved in the dial.
I'm not sure why a device in the middle of a 'normal' dialling operation
(regardless of it being either the caller/peer) would switch its SSRC
rapidly in such a fashion. A pcap should show the changes in SSRC and
might illustrate what's occurring.
Matt
Hello,
I don't think it's related to the IP-phone because I notice my
Asterisk-server also gets these messages from my SIP-provider.
The call goes : IPphone --> Asterisk --> SIP-provider
It does not occur always when calling from the same IP-phone. It can be
any IP-phone and phone type. It can also occur at any time : when there
are few calls and when there are many calls.
The negative side when this occurs is that there is no audio when the
calls gets answered. These messages flood the CLI untill the call gets
answered. Then it stops, but there is no-way-audio.
I have a second Asterisk-server (same version : 1.8.12.2) and there I
see that this messages occurs just 1 time in a call.
Could it be an issue of Asterisk ? Timing issue ? Any idea which issue
and how to tune it ?
Kind regards,
Jonas.
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