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- 488 Not acceptable when IP550 receives a Re-invite

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09-15-2011 02:26 AM
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
Solved! Go to Solution.
Accepted Solutions
09-16-2011 10:32 AM
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
09-15-2011 04:36 PM
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

09-16-2011 10:32 AM
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
02-10-2012 02:11 PM
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
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