IPOCC Dialer 488 Not Acceptable Here

[IPOCraft]

Hi there...

I was testing the Dialer in the IPOCC and my first days works fine. But then, days latter, the Campaign Monitor says "Destination Busy". When I monitor de CHAP I see 488 SIP Response "Not Acceptable Here". I'm using a SIP trunk.

The Codecs are fine (PCM 8000 ALAW) in both sides. Trace lines above ("domain.com" is the domain in SIP Trunk configuration in IPO for 10.0.11.10):

SIP Rx: TLS 10.0.11.12:55555 -> 10.0.11.10:5061
INVITE sips:871508305@domain.com SIP/2.0
From: <sips:6500@domain.com>;tag=-577dc1d65e077176723e6840_F10.0.11.12
To: <sips:871508305@domain.com>
Call-ID: 55_1436f6c5-592825a1723e6840_I@10.0.11.12
CSeq: 85 INVITE
Max-Forwards: 70
Via: SIP/2.0/TLS 10.0.11.12:5100;branch=z9hG4bK55_1436f6c5528f0091723e6882_I
Supported: replaces
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,UPDATE
User-Agent: Avaya CIE/IPOCC 10.1.2.2
Contact: <sips:6500@10.0.11.12:5100;transport=tls>
Accept-Language: en
User-To-User: 00FA08000000553ad3e14d ;encoding=hex
Content-Type: application/sdp
Content-Length: 349

v=0
o=sips:6500@10.0.11.12 1 86 IN IP4 10.0.11.12
s=sips:6500@10.0.11.12
c=IN IP4 10.0.11.12
b=TIAS:64000
b=AS:64
b=CT:64
t=0 0
m=audio 49448 RTP/SAVP 8 0 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inlinespl+m2rHBMi6766IbsgGXzwu+vkgdKVIouju/U8 UNENCRYPTED_SRTCP

CMCallEvt: 0000000000000000 0.5228.0 -1 BaseEP: NEW CMEndpoint f73f0798 TOTAL NOW=1 CALL_LIST=0
CMCallEvt: 0000000000000000 0.5228.0 -1 Contact Center.-1: NEW CMExtnEndpoint f73f0798, Name=Contact Center, Extn=6500, Phys Extn=6500
CMMap: IP::SetCodec pcp[467]b0r0 0 -> f739f558
CMCallEvt: CREATE CALL:1441 (f73f2ac8)
CMCallEvt: 0000000000000000 0.5229.0 -1 BaseEP: NEW CMEndpoint f73cc868 TOTAL NOW=2 CALL_LIST=0
CMExtnEvt: Contact Center: CMExtnHandler::SetCurrent( id: 0->5228 )

SIP Tx: TLS 10.0.11.10:5061 -> 10.0.11.12:55555
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 10.0.11.12:5100;branch=z9hG4bK55_1436f6c5528f0091723e6882_I
From: <sips:6500@domain.com>;tag=-577dc1d65e077176723e6840_F10.0.11.12
Call-ID: 55_1436f6c5-592825a1723e6840_I@10.0.11.12
CSeq: 85 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
Supported: timer,100rel
Server: IP Office 10.1.0.6.0 build 7
To: <sips:871508305@domain.com>;tag=477c8b210f9da06a
Content-Length: 0

SIP Tx: TLS 10.0.11.10:5061 -> 10.0.11.12:55555
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS 10.0.11.12:5100;branch=z9hG4bK55_1436f6c5528f0091723e6882_I
From: <sips:6500@doamin.com>;tag=-577dc1d65e077176723e6840_F10.0.11.12
Call-ID: 55_1436f6c5-592825a1723e6840_I@10.0.11.12
CSeq: 85 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
Supported: timer,100rel
Server: IP Office 10.1.0.6.0 build 7
To: <sips:871508305@doamain.com>;tag=477c8b210f9da06a
Content-Length: 0

CMExtnRx: v=6500, p1=0
CMReleaseComp
Line: type=SIPLine 360 Call: lid=385 id=5228 in=0
Cause=79, Service or option not implemented, unspecified
​

I note that the extensions calls progress fine for the same number, establishing session with SBC.

SIP Tx: UDP 10.0.11.10:5060 -> 172.16.164.218:5060
INVITE sip:59171508305@172.16.164.218 SIP/2.0
Via: SIP/2.0/UDP 10.0.11.10:5060;rport;branch=z9hG4bKcd48114bbcbdaa142dd881b4f89f7e9d
From: "2026 S SRVDORES" <sip:2026@10.0.11.10>;tag=398dde750314e97b
To: <sip:59171508305@172.16.164.218>
Call-ID: f98c9f97634536d67cff4f64462702c3
CSeq: 1925180409 INVITE
Contact: "2026 S SRVDORES" <sip:2026@10.0.11.10:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 10.1.0.6.0 build 7
P-Asserted-Identity: "2026 S SRVDORES" <sip:2026@10.0.11.10:5060>
Content-Type: application/sdp
Content-Length: 270

v=0
o=UserA 903009110 1604220327 IN IP4 10.0.11.10
s=Session SDP
c=IN IP4 10.0.11.10
t=0 0
m=audio 40762 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

SIP Rx: UDP 172.16.164.218:5060 -> 10.0.11.10:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.11.10:5060;branch=z9hG4bKcd48114bbcbdaa142dd881b4f89f7e9d;rport=5060
Call-ID: f98c9f97634536d67cff4f64462702c3
From: "2026 S SRVDORES"<sip:75819555@10.0.11.10:5060>;tag=398dde750314e97b
To: <sip:59171508305@172.16.164.218>
CSeq: 1925180409 INVITE
Content-Length: 0

CMLineRx: v=0
CMProceeding
Line: type=SIPLine 11 Call: lid=11 id=1051 in=0
Called[] Type=Default (100) Reason=CMDRdirect Calling[2026] Type=Internal Plan=Default
IE CMIERespondingPartyNumber (230)(P:100 S:100 T:0 N:100 R:4) number=871508305
IE CMIEDeviceDetail (231) 0a000b0a0000041b LOCALE=esp HW=11 VER=10 class=CMDeviceSIPTrunk type=0 number=11 channel=1 features=0x1 rx_gain=32 tx_gain=32
ep_callid=1051 ipaddr=10.0.11.10 apps=0 loc=999 em_a_loc=999 em_d_loc=0 features2=0x0 is_spcall=1 ignores_dtmf=0 avgsid=
CMARS: LINE ep Received: CMProceeding - child->state = CMCSOffering - ARS Call State = CMCSOverlapRecv
CMCallEvt: 0a000b0a0000041b 11.1051.0 21 SIPTrunk Endpoint: StateChange: END=child CMCSOffering->CMCSAccept
SIP Rx: UDP 172.16.164.218:5060 -> 10.0.11.10:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.11.10:5060;branch=z9hG4bKcd48114bbcbdaa142dd881b4f89f7e9d;rport=5060
Record-Route: <sip:172.16.164.218:5060;transport=udp;lr;Hpt=8eb8_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=9876>
Call-ID: f98c9f97634536d67cff4f64462702c3
From: "2026 S SRVDORES"<sip:75819555@10.0.11.10:5060>;tag=398dde750314e97b
To: <sip:59171508305@172.16.164.218>;tag=a1q0x1xj-CC-1013-OFC-422
CSeq: 1925180409 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
Contact: <sip:172.16.164.218:5060;transport=udp;Hpt=8eb8_16;CxtId=3;TRC=ffffffff-ffffffff>
Content-Length: 196
Content-Type: application/sdp

v=0
o=- 1127339199 1127339199 IN IP4 172.16.164.218
s=SBC call
c=IN IP4 172.16.164.218
t=0 0
m=audio 28922 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20​

Some ideas, please???

 

[Alfalis]

Without knowing your configuration it is hard to tell what might be the issue.
What comes up when you enable "SIP -> Verbose" (SIP tab) in SysMon?