Solved: 488 Not acceptable when IP550 receives a Re-invite - HP Support Community

Anonymous · ‎09-15-2011

Hello,
I've configured 2 Polycom SoundPoint IP550 to work with a Genesys SIP Server.

those phones are supposed to be Agents phones in a call center.
It seems like the phone does not accepte to be reinvited by the SIPServer

I have the following environment :
2 polycom :

IP : 10.140.4.53 and 10.140.4.54

Model IP550

Assembly 2345-12500-001 Rev R

BootBlock 2.8.1 (12500_001)

BootRom 4.3.0.0246

Application SIP : 3.2.5.0508

P/N: 3150-11530-325

Components Label PolyDSP Titan Mem1FS3 (G729)

version 3.1.6.0004

P/N: 3150-11580-316

the servers :

SIP Server : 10.140.4.891

Stream Manager : 10.140.4.85

attached my config files.

hereunder the SIP messages taken from the SIPServer logs :

##### Polycom INVITES 123 on the SIPServer #####

09:56:54.873 Received [472,UDP] 852 bytes from 10.140.4.54:5060 <<<<<
INVITE sip:123@10.140.4.89:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.140.4.54:5060;branch=z9hG4bK2aaa32cfC022DA6
From: "OBW_2154" <sip:2154@10.140.4.89>;tag=2D762945-EF67D1D4
To: <sip:123@10.140.4.89;user=phone>
CSeq: 1 INVITE
Call-ID: a8807489-6bf8e968-1d6a8f3@10.140.4.54
Contact: <sip:2154@10.140.4.54:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 221

v=0
o=- 1316070606 1316070606 IN IP4 10.140.4.54
s=Polycom IP Phone
c=IN IP4 10.140.4.54
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 8 0 127
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000

###### SIP Server replies RINGING ######

09:56:58.452 Sending  [472,UDP] 445 bytes to 10.140.4.54:5060 >>>>>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.140.4.54:5060;branch=z9hG4bK2aaa32cfC022DA6;received=10.140.4.54
From: "OBW_2154" <sip:2154@10.140.4.89>;tag=2D762945-EF67D1D4
To: <sip:123@10.140.4.89;user=phone>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-237
CSeq: 1 INVITE
Call-ID: a8807489-6bf8e968-1d6a8f3@10.140.4.54
X-Genesys-CallUUID: 8Q29Q1U4U14M76B6FC1C2B4R2K00007M
Allow: INVITE, ACK, PRACK, CANCEL, BYE, REFER, INFO
Content-Length: 0

#### SIP SERver invites another resources (StreamManager) to play an audio file in Netann (RTP Stream) ####

INVITE sip:annc@10.140.4.85:5062;play=announcement/EN/welcome_EN;repeat=1 SIP/2.0
From: <sip:2154@10.140.4.89:5060>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-238
To: <sip:annc@10.140.4.85:5062;play=announcement/EN/welcome_EN;repeat=1>
Call-ID: BA76BB0B-F947-467F-96B8-73B1CB6C720B-19766@10.140.4.89
CSeq: 1 INVITE
Content-Length: 212
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.140.4.89:5060;branch=z9hG4bK50420631-0DD7-49D6-A9E5-C96535D16EBB-261
Contact: <sip:2154@10.140.4.89:5060>
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER
X-Genesys-CallUUID: 8Q29Q1U4U14M76B6FC1C2B4R2K00007M
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Allow-Events: talk,hold,conference
Max-Forwards: 69
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: 100rel,timer

v=0
o=- 1316003250 1 IN IP4 10.140.4.54
s=Polycom IP Phone
c=IN IP4 10.140.4.54
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 8 0 127
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000

#### Stream Manager replies 200OK ####

09:57:03.858 Received [472,UDP] 732 bytes from 10.140.4.85:4408 <<<<<
SIP/2.0 200 OK
From: <sip:2154@10.140.4.89:5060>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-238
To: <sip:annc@10.140.4.85:5062;play=announcement/EN/welcome_EN;repeat=1>;tag=5F601A49-23D1-4FA3-BCFC-614DAA3D9F90-175
Call-ID: BA76BB0B-F947-467F-96B8-73B1CB6C720B-19766@10.140.4.89
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.140.4.89:5060;branch=z9hG4bK50420631-0DD7-49D6-A9E5-C96535D16EBB-261;received=10.140.4.89
Contact: <sip:10.140.4.85:5062>
Content-Type: application/sdp
Content-Length: 238

v=0
o=Genesys 107 107 IN IP4 10.140.4.85
s=StreamManager 7.6.002.02 play
c=IN IP4 10.140.4.85
t=0 0
m=audio 8350 RTP/AVP 8 127
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
a=sendrecv
a=rtpmap:8 pcma/8000
a=x-media-op:play

##### SIP SErver Send 200OK to the polycom ####

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.140.4.54:5060;branch=z9hG4bK2aaa32cfC022DA6;received=10.140.4.54
From: "OBW_2154" <sip:2154@10.140.4.89>;tag=2D762945-EF67D1D4
To: <sip:123@10.140.4.89;user=phone>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-237
CSeq: 1 INVITE
Call-ID: a8807489-6bf8e968-1d6a8f3@10.140.4.54
Contact: <sip:10.140.4.89:5060>
X-Genesys-CallUUID: 8Q29Q1U4U14M76B6FC1C2B4R2K00007M
Allow: INVITE, ACK, PRACK, CANCEL, BYE, REFER, INFO
Session-Expires: 1800;refresher=uas
Min-SE: 90
Supported: timer
Content-Type: application/sdp
Content-Length: 227

v=0
o=Genesys 1316003245 1 IN IP4 10.140.4.85
s=StreamManager
c=IN IP4 10.140.4.85
t=0 0
m=audio 8350 RTP/AVP 8 127
a=sendrecv
a=x-media-op:play
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
a=rtpmap:8 pcma/8000

#### Polycom ACK the 200OK#####

09:57:03.920 Received [472,UDP] 559 bytes from 10.140.4.54:5060 <<<<<
ACK sip:10.140.4.89:5060 SIP/2.0
Via: SIP/2.0/UDP 10.140.4.54:5060;branch=z9hG4bKcba099ee45B5FC71
From: "OBW_2154" <sip:2154@10.140.4.89>;tag=2D762945-EF67D1D4
To: <sip:123@10.140.4.89;user=phone>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-237
CSeq: 1 ACK
Call-ID: a8807489-6bf8e968-1d6a8f3@10.140.4.54
Contact: <sip:2154@10.140.4.54:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

#### ACK is forwarded to StreamManager #####

09:57:03.936 Sending  [472,UDP] 552 bytes to 10.140.4.85:5062 >>>>>
ACK sip:10.140.4.85:5062 SIP/2.0
From: <sip:2154@10.140.4.89:5060>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-238
To: <sip:annc@10.140.4.85:5062;play=announcement/EN/welcome_EN;repeat=1>;tag=5F601A49-23D1-4FA3-BCFC-614DAA3D9F90-175
Call-ID: BA76BB0B-F947-467F-96B8-73B1CB6C720B-19766@10.140.4.89
CSeq: 1 ACK
Content-Length: 0
Via: SIP/2.0/UDP 10.140.4.89:5060;branch=z9hG4bK50420631-0DD7-49D6-A9E5-C96535D16EBB-262
Allow: INVITE, ACK, PRACK, CANCEL, BYE, REFER, INFO
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en

#### Once the audio file is played, Stream Manager sends a BYE to SIP Server ###

09:57:06.217 Received [472,UDP] 449 bytes from 10.140.4.85:4408 <<<<<
BYE sip:2154@10.140.4.89:5060 SIP/2.0
From: <sip:annc@10.140.4.85:5062;play=announcement/EN/welcome_EN;repeat=1>;tag=5F601A49-23D1-4FA3-BCFC-614DAA3D9F90-175
To: <sip:2154@10.140.4.89:5060>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-238
Call-ID: BA76BB0B-F947-467F-96B8-73B1CB6C720B-19766@10.140.4.89
CSeq: 1 BYE
Content-Length: 0
Via: SIP/2.0/UDP 10.140.4.85:5062;branch=z9hG4bKE2724573-6503-43DD-AF7D-C6A1922A9688-43
Reason: SIP;cause=200

### SIPSServer replies OK to the BYE ####

09:57:06.217 Sending  [472,UDP] 569 bytes to 10.140.4.85:5062 >>>>>
SIP/2.0 200 OK
From: <sip:annc@10.140.4.85:5062;play=announcement/EN/welcome_EN;repeat=1>;tag=5F601A49-23D1-4FA3-BCFC-614DAA3D9F90-175
To: <sip:2154@10.140.4.89:5060>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-238
Call-ID: BA76BB0B-F947-467F-96B8-73B1CB6C720B-19766@10.140.4.89
CSeq: 1 BYE
Via: SIP/2.0/UDP 10.140.4.85:5062;branch=z9hG4bKE2724573-6503-43DD-AF7D-C6A1922A9688-43;received=10.140.4.85
Contact: <sip:2154@10.140.4.89:5060>
X-Genesys-CallUUID: 8Q29Q1U4U14M76B6FC1C2B4R2K00007M
Allow: INVITE, ACK, PRACK, CANCEL, BYE, REFER, INFO
Content-Length: 0

#### The, SIP Server request another audio file to be played by the Stream Manager ###

09:57:06.233 Sending  [472,UDP] 692 bytes to 10.140.4.85:5062 >>>>>
INVITE sip:annc@10.140.4.85:5062;play=announcement/SE/welcome_SE;repeat=1 SIP/2.0
From: <sip:2154@10.140.4.89:5060>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-239
To: <sip:annc@10.140.4.85:5062;play=announcement/SE/welcome_SE;repeat=1>
Call-ID: BA76BB0B-F947-467F-96B8-73B1CB6C720B-19770@10.140.4.89
CSeq: 1 INVITE
Content-Length: 0
Via: SIP/2.0/UDP 10.140.4.89:5060;branch=z9hG4bK50420631-0DD7-49D6-A9E5-C96535D16EBB-263
Contact: <sip:2154@10.140.4.89:5060>
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER
X-Genesys-CallUUID: 8Q29Q1U4U14M76B6FC1C2B4R2K00007M
Max-Forwards: 69
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: 100rel,timer

#### Stream Manager replies OK ###

09:57:06.389 Received [472,UDP] 720 bytes from 10.140.4.85:4408 <<<<<
SIP/2.0 200 OK
From: <sip:2154@10.140.4.89:5060>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-239
To: <sip:annc@10.140.4.85:5062;play=announcement/SE/welcome_SE;repeat=1>;tag=5F601A49-23D1-4FA3-BCFC-614DAA3D9F90-176
Call-ID: BA76BB0B-F947-467F-96B8-73B1CB6C720B-19770@10.140.4.89
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.140.4.89:5060;branch=z9hG4bK50420631-0DD7-49D6-A9E5-C96535D16EBB-263;received=10.140.4.89
Contact: <sip:10.140.4.85:5062>
Content-Type: application/sdp
Content-Length: 226

v=0
o=Genesys 108 108 IN IP4 10.140.4.85
s=StreamManager 7.6.002.02 play
c=IN IP4 10.140.4.85
t=0 0
m=audio 8352 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 pcma/8000
a=x-media-op:play

### An RE-invite is sent to the polycom by the SIP SErver###

09:57:06.389 Sending  [472,UDP] 848 bytes to 10.140.4.54:5060 >>>>>
INVITE sip:2154@10.140.4.54:5060 SIP/2.0
From: <sip:123@10.140.4.89;user=phone>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-237
To: "OBW_2154" <sip:2154@10.140.4.89>;tag=2D762945-EF67D1D4
Call-ID: a8807489-6bf8e968-1d6a8f3@10.140.4.54
CSeq: 1 INVITE
Content-Length: 220
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.140.4.89:5060;branch=z9hG4bK50420631-0DD7-49D6-A9E5-C96535D16EBB-264
Contact: <sip:10.140.4.89:5060>
Allow: INVITE, ACK, PRACK, CANCEL, BYE, REFER, INFO
X-Genesys-CallUUID: 8Q29Q1U4U14M76B6FC1C2B4R2K00007M
Max-Forwards: 70
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: 100rel,timer

v=0
o=Genesys 1316003245 3 IN IP4 0.0.0.0
s=StreamManager 7.6.002.02 play
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 8 101
a=x-media-op:play
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 pcma/8000

### SIP SErver replies 488 Not Acceptable Here ###

09:57:06.405 Received [472,UDP] 469 bytes from 10.140.4.54:5060 <<<<<
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.140.4.89:5060;branch=z9hG4bK50420631-0DD7-49D6-A9E5-C96535D16EBB-264
From: <sip:123@10.140.4.89;user=phone>;tag=7411F039-4D10-46F7-ADB5-C493B3A2C2F1-237
To: "OBW_2154" <sip:2154@10.140.4.89>;tag=2D762945-EF67D1D4
CSeq: 1 INVITE
Call-ID: a8807489-6bf8e968-1d6a8f3@10.140.4.54
Contact: <sip:2154@10.140.4.54:5060>
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508
Accept-Language: en
Content-Length: 0

I also have the same 488 problem when I configured callForward on a polycom and call it with another polycom. the polycom caller does not accept the re-invite, call is terminated without beeing forwarded.

I've look throught the Admin guide to find an config paramter in sip.cfg or phone1.cfg with no success.

This Scenario works perfectly with all the other SIP Phones i've been working (mostly softphones)

Thanks for your help

SteffenBaierUK · ‎09-16-2011

Hello theduf,

Genesys should be able to provide you with provisioning templates on how to configure a Polycom Phone on their Platform as they would have done the Interop Testing.

Looking at the Information you provided and the MAC Address of the Phone shows the Unit is still within warranty is the reason why I suggested to work with your Reseller and/or Polycom Support to check the information and a trace if provided.

Another community Member may have experience with your setup and may reply.

Best Regards

Steffen Baier

Polycom Global Services

------------------------------------------------
Notice: This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your support channels.
Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN

View solution in original post

SteffenBaierUK · ‎09-15-2011

Hello theduf,

welcome to the Polycom Community.

The Forum is not really the correct place to report an issue and looking at the Phones MAC Address you should work with your Reseller or Polycom Support directly in order to check for the root cause as explained in more detail => here <=.

Please provide an unfiltered Wireshark Trace, all configuration files used to Provision the Phone and an -app and -boot.log when opening a ticket.

Best Regards

Steffen Baier

Polycom Global Services

------------------------------------------------
Notice: This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your support channels.
Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN

Anonymous · ‎09-16-2011

Hello,

I'm not reporting an issue, I'm asking help in order to configure my polycom phone correctly...

If I can't ask for help here, what is this forum for ?

Best Regards,

SteffenBaierUK · ‎09-16-2011

Hello theduf,

Genesys should be able to provide you with provisioning templates on how to configure a Polycom Phone on their Platform as they would have done the Interop Testing.

Looking at the Information you provided and the MAC Address of the Phone shows the Unit is still within warranty is the reason why I suggested to work with your Reseller and/or Polycom Support to check the information and a trace if provided.

Another community Member may have experience with your setup and may reply.

Best Regards

Steffen Baier

Polycom Global Services

------------------------------------------------
Notice: This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your support channels.
Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN

Chad_A · ‎02-10-2012

Hello,

Per RFC 3264 - The re-INVITE you are sending is out of the RFC spec for RFC 2543 hold;

RFC 2543 [10] specified that placing a user on hold was accomplished
by setting the connection address to 0.0.0.0. Its usage for putting
a call on hold is no longer recommended, since it doesn't allow for
RTCP to be used with held streams, doesn't work with IPv6, and breaks
with connection oriented media. However, it can be useful in an
initial offer when the offerer knows it wants to use a particular set
of media streams and formats, but doesn't know the addresses and
ports at the time of the offer. Of course, when used, the port
number MUST NOT be zero, which would specify that the stream has been
disabled. An agent MUST be capable of receiving SDP with a
connection address of 0.0.0.0, in which case it means that neither
RTP nor RTCP should be sent to the peer.

So the phone should send a 488 in this scenario.

-Chad

488 Not acceptable when IP550 receives a Re-invite

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