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<font face="Helvetica, Arial, sans-serif">Hello,<br>
<br>
I have the following construction :<br>
<br>
<br>
Provider --> SipAgent (asterisk) --> Asterisk Server_A
--> IP-phone (Snom 370)<br>
<br>
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.<br>
<br>
My SipAgent has DirectMedia=yes so there is no audio flowing
through this SipAgent. It only stays in the signaling path (SIP).<br>
<br>
My SipAgent will communicate in a SIP re-INVITE the audio ports of
the Asterisk Server_A to the "Provider".<br>
My SipAgent will communicate in a SIP re-INVITE the audio ports of
the "Provider" to the Asterisk Server_A.<br>
Audio will flow directly between "Provider" and "Asterisk
Server_A".<br>
<br>
This works great.<br>
<br>
<br>
On my Asterisk Server_A, I see the following :<br>
<br>
<i>SIP/SipAgent-00000bf9 requested media update control 26,
passing it to SIP/ead14-00000bfb</i><br>
<br>
Mostly this appears one time in a call. This I find normal.<br>
<br>
But sometimes the CLI is flooded with 100 of these messages... and
that I find NOT NORMAL.<br>
<br>
The flood stops when the call is anwered.<br>
<br>
<br>
<br>
This is the SIP INVITE on my SipAgent :<br>
<br>
INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:xx32xxxxxx@XX.XX.XX.199:5060">sip:xx32xxxxxx@XX.XX.XX.199:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP XX.XX.XX.198:5060;branch=z9hG4bK37fc69a2;rport<br>
Max-Forwards: 70<br>
From: "</font><font face="Helvetica, Arial, sans-serif"><font
face="Helvetica, Arial, sans-serif">xx35xxxxxx</font>" <sip:</font><font
face="Helvetica, Arial, sans-serif"><font face="Helvetica, Arial,
sans-serif">xx35xxxxxx</font>@XX.XX.XX.198>;tag=as3bbe54ca<br>
To: <sip:</font><font face="Helvetica, Arial, sans-serif"><font
face="Helvetica, Arial, sans-serif">xx32xxxxxx</font>@XX.XX.XX.199>;tag=as180f6a04<br>
Contact: <sip:</font><font face="Helvetica, Arial, sans-serif"><font
face="Helvetica, Arial, sans-serif"><font face="Helvetica,
Arial, sans-serif">xx35xxxxxx</font></font>@XX.XX.XX.198:5060><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:675c1f3f5141f5ac0e981c27414de0be@XX.XX.XX.198:5060">675c1f3f5141f5ac0e981c27414de0be@XX.XX.XX.198:5060</a><br>
CSeq: 103 INVITE<br>
User-Agent: PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>
X-asterisk-Info: SIP re-invite (External RTP bridge)<br>
Content-Type: application/sdp<br>
Content-Length: 239<br>
<br>
<br>
X-Asterisk-Info shows the RTP bridge, which I find normal.<br>
<br>
And my Asterisk Server_A answers with "100 Trying".<br>
<br>
<br>
Now, what could be the difference between a call where the CLI on
Asterisk Server_A tells </font><font face="Helvetica, Arial,
sans-serif"><i>requested media update control 26</i> one time and
where it floods the CLI ?<i><br>
</i><br>
<br>
Kind regards,<br>
<br>
Jonas.<br>
<i><br>
</i></font>
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