<div dir="ltr"><div class="gmail_extra"><br><div class="gmail_quote">On Sun, Jan 11, 2015 at 11:19 PM, Michael Englehorn <span dir="ltr"><<a href="mailto:michael@englehorn.com" target="_blank">michael@englehorn.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">-----BEGIN PGP SIGNED MESSAGE-----<br>
Hash: SHA1<br>
<br>
Is it possible to use the instant messaging feature of Polycom phones in<br>
Asterisk? At the moment I'm seeing this in the SIP messaging when I try<br>
to send one from a Polycom 450.<br>
<br>
<--- SIP read from UDP:<CENSORED POLYCOM IP>:5060 ---><br>
INVITE sip:0100@<CENSORED>:5060;user=phone SIP/2.0<br>
Via: SIP/2.0/UDP <CENSORED POLYCOM IP>;branch=z9hG4bK484dcd1fDD872ECE<br>
From: "Michael" <sip:3109@<CENSORED>>;tag=D2DAE96E-D8618427<br>
To: <sip:0100@<CENSORED>;user=phone><br>
CSeq: 2 INVITE<br>
Call-ID: d2c5011e-1d718f7-9203b936@<CENSORED POLYCOM IP><br>
Contact: <sip:3109@<CENSORED POLYCOM IP>><br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,<br>
NOTIFY, PRACK, UPDATE, REFER<br>
User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.7.2514<br>
Accept-Language: en<br>
Supported: 100rel,replaces<br>
Allow-Events: conference,talk,hold<br>
Authorization: Digest username="3109", realm="asterisk",<br>
nonce="<CENSORED>", uri="sip:0100@<CENSORED>:5060;user=phone",<br>
response="<CENSORED>", algorithm=MD5<br>
Max-Forwards: 70<br>
Content-Type: application/sdp<br>
Content-Length: 143<br>
<br>
v=0<br>
o=- 1421039199 1421039199 IN IP4 <CENSORED POLYCOM IP><br>
s=Polycom IP Phone<br>
c=IN IP4 <CENSORED POLYCOM IP><br>
t=0 0<br>
m=message 5060 sip sip:3109@<CENSORED><br>
<-------------><br>
SIP/2.0 488 Not acceptable here<br>
Via: SIP/2.0/UDP <CENSORED POLYCOM<br>
IP>;branch=z9hG4bK484dcd1fDD872ECE;received=<CENSORED POLYCOM<br>
IP>;rport=5060<br>
From: "Michael" <sip:3109@<CENSORED>>;tag=D2DAE96E-D8618427<br>
To: <sip:0100@<CENSORED>;user=phone>;tag=as3d0d8c04<br>
Call-ID: d2c5011e-1d718f7-9203b936@<CENSORED POLYCOM IP><br>
CSeq: 2 INVITE<br>
Server: FPBX-2.11.0(11.9.0)<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,<br>
INFO, PUBLISH, MESSAGE<br>
Supported: replaces, timer<br>
Content-Length: 0<br>
<br clear="all"></blockquote><div><br></div><div>Asterisk does not understand or support an SDP media type of 'message'.<br><br></div><div>Both chan_pjsip and chan_sip can support SIP MESSAGE requests, received both in dialog and out of dialog. In addition, chan_sip will handle media types of 'text' for real-time text received in the RTP stream.<br></div><br></div>-- <br><div class="gmail_signature"><div dir="ltr"><div>Matthew Jordan<br></div><div>Digium, Inc. | Engineering Manager</div><div>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA</div><div>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></div></div></div>
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