[asterisk-users] Odd Issue With Polycom Phones]

[Gord Urquhart]

The phone is only making one call, notice the call-id did not change.
The second INVITE is sent in responce to a 401 Authentication
Required. The 401 will contain the necessary authentication
information for the phone to use to build the Authorization header
that it inserts in the second invite. THe mechanism uses a shared
secret (the reg.X.auth.userId and reg.X.auth.password in the polycom
cfg file, and the secret="XXXXX" and the userID(I think thats what its
called) in the asterisk config files).

If you have other phones that are not doing this second invite I would
bet its because on the asterisk side you have not configured them to
use a secret.

----------------------------------------------------------------------------------
Thanks for the tip, I did just that, and now I am more confused.

It does appear as though there is just one call ID (if my assumption
that the "tag=" determines the call.

The first time it sends like this:

<--- SIP read from UDP:x.x.x.x:5060 --->
INVITE sip:3261 at y.y.y.y;user=phone SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKe3e15c76913F8BDD
From: "3271" <sip:3271@ y.y.y.y > <sip:3271 at y.y.y.y>;tag=990EE6B0-8E3DCEA7
To: <sip:3261@ y.y.y.y;user=phone> <sip:3261 at y.y.y.y;user=phone>
CSeq: 1 INVITE
Call-ID: 96a1fe9c-88f06c73-7e209322 at x.x.x.x
Contact: <sip:3271@ x.x.x.x:5060> <sip:3271 at x.x.x.x:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 461

v=0
o=- 1271881915 1271881915 IN IP4 x.x.x.x
s=Polycom IP Phone
c=IN IP4 x.x.x.x
t=0 0
a=sendrecv
m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:99 SIREN14/16000
a=fmtp:99 bitrate=48000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

Asterisk responds with a SIP/2.0 401 Unauthorized, the phone then
comes back with this:

<--- SIP read from UDP:x.x.x.x:5060 --->
INVITE sip:3261@ y.y.y.y;user=phone SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK6f7a6692AF94008
From: "3271" <sip:3271@ y.y.y.y > <sip:3271 at y.y.y.y>;tag=990EE6B0-8E3DCEA7
To: <sip:3261@ y.y.y.y;user=phone> <sip:3261 at y.y.y.y;user=phone>
CSeq: 2 INVITE
Call-ID: 96a1fe9c-88f06c73-7e209322@ x.x.x.x
Contact: <sip:3271@ x.x.x.x:5060> <sip:3271 at x.x.x.x:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Authorization: Digest username="3271", realm="asterisk",
nonce="393a1b1f", uri="sip:3261@ y.y.y.y;user=phone"
<sip:3261 at y.y.y.y;user=phone>,
response="c8223e261c252c12172982ee661ad307", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 461

v=0
o=- 1271881915 1271881915 IN IP4 x.x.x.x
s=Polycom IP Phone
c=IN IP4 x.x.x.x
t=0 0
a=sendrecv
m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:99 SIREN14/16000
a=fmtp:99 bitrate=48000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000


The difference is that the CSeq is now 2 and the following line is added:

Authorization: Digest username="3271", realm="asterisk",
nonce="393a1b1f", uri="sip:3261 at y.y.y.y;user=phone"
<sip:3261 at y.y.y.y;user=phone>,
response="c8223e261c252c12172982ee661ad307", algorithm=MD5


So maybe I do have an issue in Asterisk, okay probably.  Any clues as
to how to debug?  Let me know if need to post more information.

Thanks.

-Jay

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com
<asterisk-users-bounces at lists.digium.com>] On Behalf Of Sean Brady
Sent: Tuesday, April 20, 2010 4:57 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Odd Issue With Polycom Phones
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