[asterisk-users] Polycom instant messages

[Matthew Jordan]

On Sun, Jan 11, 2015 at 11:19 PM, Michael Englehorn <michael at englehorn.com>
wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Is it possible to use the instant messaging feature of Polycom phones in
> Asterisk? At the moment I'm seeing this in the SIP messaging when I try
> to send one from a Polycom 450.
>
> <--- SIP read from UDP:<CENSORED POLYCOM IP>:5060 --->
> INVITE sip:0100@<CENSORED>:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP <CENSORED POLYCOM IP>;branch=z9hG4bK484dcd1fDD872ECE
> From: "Michael" <sip:3109@<CENSORED>>;tag=D2DAE96E-D8618427
> To: <sip:0100@<CENSORED>;user=phone>
> CSeq: 2 INVITE
> Call-ID: d2c5011e-1d718f7-9203b936@<CENSORED POLYCOM IP>
> Contact: <sip:3109@<CENSORED POLYCOM IP>>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.7.2514
> Accept-Language: en
> Supported: 100rel,replaces
> Allow-Events: conference,talk,hold
> Authorization: Digest username="3109", realm="asterisk",
> nonce="<CENSORED>", uri="sip:0100@<CENSORED>:5060;user=phone",
> response="<CENSORED>", algorithm=MD5
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 143
>
> v=0
> o=- 1421039199 1421039199 IN IP4 <CENSORED POLYCOM IP>
> s=Polycom IP Phone
> c=IN IP4 <CENSORED POLYCOM IP>
> t=0 0
> m=message 5060 sip sip:3109@<CENSORED>
> <------------->
> SIP/2.0 488 Not acceptable here
> Via: SIP/2.0/UDP <CENSORED POLYCOM
> IP>;branch=z9hG4bK484dcd1fDD872ECE;received=<CENSORED POLYCOM
> IP>;rport=5060
> From: "Michael" <sip:3109@<CENSORED>>;tag=D2DAE96E-D8618427
> To: <sip:0100@<CENSORED>;user=phone>;tag=as3d0d8c04
> Call-ID: d2c5011e-1d718f7-9203b936@<CENSORED POLYCOM IP>
> CSeq: 2 INVITE
> Server: FPBX-2.11.0(11.9.0)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
>
>
Asterisk does not understand or support an SDP media type of 'message'.

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.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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