Hello,
I have the following construction :
Provider --> SipAgent (asterisk) --> Asterisk Server_A --> IP-phone
(Snom 370)
If a call comes in from the "Provider" to my SipAgent, then my SipAgent
send the call to the correct Asterisk Server_A (dialplan logic based on
number). The Asterisk Server_A takes the call and sends it to the IP-phone.
My SipAgent has DirectMedia=yes so there is no audio flowing through
this SipAgent. It only stays in the signaling path (SIP).
My SipAgent will communicate in a SIP re-INVITE the audio ports of the
Asterisk Server_A to the "Provider".
My SipAgent will communicate in a SIP re-INVITE the audio ports of the
"Provider" to the Asterisk Server_A.
Audio will flow directly between "Provider" and "Asterisk Server_A".
This works great.
On my Asterisk Server_A, I see the following :
/SIP/SipAgent-00000bf9 requested media update control 26, passing it to
SIP/ead14-00000bfb/
Mostly this appears one time in a call. This I find normal.
But sometimes the CLI is flooded with 100 of these messages... and that
I find NOT NORMAL.
The flood stops when the call is anwered.
This is the SIP INVITE on my SipAgent :
INVITE sip:xx32xxx...@xx.xx.xx.199:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.198:5060;branch=z9hG4bK37fc69a2;rport
Max-Forwards: 70
From: "xx35xxxxxx" <sip:xx35xxx...@xx.xx.xx.198>;tag=as3bbe54ca
To: <sip:xx32xxx...@xx.xx.xx.199>;tag=as180f6a04
Contact: <sip:xx35xxx...@xx.xx.xx.198:5060>
Call-ID: 675c1f3f5141f5ac0e981c27414de...@xx.xx.xx.198:5060
CSeq: 103 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 239
X-Asterisk-Info shows the RTP bridge, which I find normal.
And my Asterisk Server_A answers with "100 Trying".
Now, what could be the difference between a call where the CLI on
Asterisk Server_A tells /requested media update control 26/ one time and
where it floods the CLI ?/
/
Kind regards,
Jonas.
/
/
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