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michael at englehorn.com Guest
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Posted: Mon Jan 12, 2015 12:19 am Post subject: [asterisk-users] Polycom instant messages |
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-----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
Thank you,
- --
Michael J. Englehorn
H: 952-884-6776 (Use first)
E-Mail: Michael@englehorn.com
- ----
Those parts of the system that you can hit with a hammer (not advised)
are called hardware; those program instructions that you can only curse
at are called software.
-- Levitating Trains and Kamikaze Genes: Technological
Literacy for the 1990's
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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mjordan at digium.com Guest
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Posted: Mon Jan 12, 2015 8:40 am Post subject: [asterisk-users] Polycom instant messages |
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On Sun, Jan 11, 2015 at 11:19 PM, Michael Englehorn <michael@englehorn.com (michael@englehorn.com)> wrote:
Quote: | -----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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asterisk at voipbusine... Guest
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Posted: Mon Jan 12, 2015 10:53 am Post subject: [asterisk-users] Polycom instant messages |
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I didn't know that instant messaging was a feature with the Polycom's. Do
you have any documentation, how-to's, etc. that you can point me to? That
would be just way too cool.
Thanks;
John
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael
Englehorn
Sent: Monday, January 12, 2015 12:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom instant messages
-----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
Thank you,
- --
Michael J. Englehorn
H: 952-884-6776 (Use first)
E-Mail: Michael@englehorn.com
- ----
Those parts of the system that you can hit with a hammer (not advised) are
called hardware; those program instructions that you can only curse at are
called software.
-- Levitating Trains and Kamikaze Genes: Technological
Literacy for the 1990's
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--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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michael at englehorn.com Guest
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Posted: Mon Jan 12, 2015 12:00 pm Post subject: [asterisk-users] Polycom instant messages |
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Yeah, I found the information on how to enable it here:
http://support.polycom.com/global/documents/support/setup_maintenance/products/voice/uc_ag_4_0_5.pdf
It's on page 117 (or page 205 according to Adobe).
It gives you an extra option in Messaging that lets you send and receive
messages.
On Mon, Jan 12, 2015 at 10:52:24AM -0500, Tech Support wrote:
Quote: | I didn't know that instant messaging was a feature with the Polycom's. Do
you have any documentation, how-to's, etc. that you can point me to? That
would be just way too cool.
Thanks;
John
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael
Englehorn
Sent: Monday, January 12, 2015 12:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom instant messages
-----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
Thank you,
- --
Michael J. Englehorn
H: 952-884-6776 (Use first)
E-Mail: Michael@englehorn.com
- ----
Those parts of the system that you can hit with a hammer (not advised) are
called hardware; those program instructions that you can only curse at are
called software.
-- Levitating Trains and Kamikaze Genes: Technological
Literacy for the 1990's
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--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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- --
Michael J. Englehorn
H: 952-884-6776
E-Mail: Michael@englehorn.com
- ----
Those parts of the system that you can hit with a hammer (not advised)
are called hardware; those program instructions that you can only curse
at are called software.
-- Levitating Trains and Kamikaze Genes: Technological
Literacy for the 1990's
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--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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