On Sun, Jan 11, 2015 at 11:19 PM, Michael Englehorn <mich...@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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