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SIP Line plays Local Music on Hold (MoH) when dialing thru AA and being put on Hold

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blue42comz

Programmer
Feb 27, 2012
14
0
0
AU
Hi All,

Have a strange fault happening at 2x sites. Both sites have Avaya IPO R9.1 with SIP trunks.

Issue:
Avaya User dials a specific external number (in this case it is a bank).
Remote party answers the call with an Auto Attendant and plays greeting with a list of options.
Avaya User dials digit to progress through the AA.
The Avaya User is immediately put on hold - but is listening to their own systems local Music on Hold (MOH).
The call is never connected back to the remote party.

Trouble shooting:
- Have replicated the issue on 2 separate Avaya IPO R9.1 systems both dialing to the same number - same result.
- Does not happen with all auto attendants.
- Called the same number from my mobile. Selected AA options - works ok.
- Configured a SIP trunk on a different system (Panasonic NS-700) dialed the same number from the Panasonic - works ok.
- Monitor traces shows new SIP INVITE messages being Rx from the remote party and then the local Avaya IPO MoH is heard.
- System Status shows the call in progress. Even when the Avaya User is hearing the local MoH.
- Have spoken with out Tech support team and the advised that R10 has a new option in VoIP tab to enable/disable "Local Hold Music".

Questions:
- Has anyone else experienced anything similar to the above?
- Does anyone have a work around for R9.1?

Any help is much appreciated!

Kind regards
 
Exactly which 9.1 version do you have?
What kind of endpoint does the user have?
What do the press in the AA?

"Trying is the first step to failure..." - Homer
 
- R9.1.500.145 & R9.1.0.437
- Various endpoint including 1416/9508/9641
- Options for the remote parties AA are "Dial 1- to make a payment. Dial 2- for all other enquiries." (Note: It does not matter which option you select. They both give same result. Avaya User is put on Hold and Hears their own systems MoH).
 
Below is monitor trace from the moment the User presses 2...
(Note: Some phone numbers and IP addresses have been partially blocked)

******************************************************************************
10:28:23 237150683mS CMExtnEvt: v=(null) State, new=Idle old=Idle,0,0,259: Digit Key Pressed 2
10:28:23 237150685mS CMLineTx: v=0
CMInformation
Line: type=SIPLine 17 Call: lid=17 id=13066 in=0
Called[2] Type=Default (100) Reason=CMDRdirect
BChan: slot=20 chan=3
IE CMIERespondingPartyName (228)(Type=CMNameDefault) name=Tech
IE CMIERespondingPartyNumber (230)(P:100 S:100 T:101 N:100 R:4) number=259
IE CMIEDeviceDetail (231) c0a8101000003306 LOCALE=ena HW=15 VER=9 class=CMDeviceStdPhone type=78 number=51 channel=0 features=0x10000420 rx_gain=32 tx_gain=32 ep_callid=13062 ipaddr=192.168.16.XXX apps=0 loc=0 em_loc=1 features2=0x0 is_spcall=0
10:28:23 237150685mS Sip: c0a810100000330a 17.13066.0 440 SIPTrunk Endpoint(f16011f4) received CMInformation
10:28:23 237150687mS CMMap: a=20.3 b=0.0 PCGS CPReserveCodec (pcp[546]b0r1) true
10:28:23 237150687mS CMMap: a=20.3 b=0.0 PCGS CPReserveCodec (pcp[546]b0r1) true
10:28:23 237150688mS CMMap: PCG::MapBChan pcp[546]b0r1 cp_b f545b1e4 other_cp_b f4fb2194 type CGTypeSimple
10:28:23 237150688mS CMMap: a=20.3 b=2.20 M12
10:28:23 237150688mS CMMap: PlatformConnectionGroupDTMF send rfc2833 inband to enc codec
10:28:23 237150690mS CMMap: a=21.43 b=0.0 DTMF::AllocateRAS allocated CMRTVoicemail(f5451de4) resource busy 3, total 45
10:28:23 237150690mS CMMap: a=21.43 b=1.65535 T
10:28:23 237150690mS CMMap: PCG::UnmapBChan pcp[95]b1r0 cp_b f4fb2194 other_cp_b f545b1e4
10:28:23 237150690mS CMMap: a=2.20 b=20.3 M02
10:28:23 237150691mS H323Evt: SESS 1a0 SetOperational local 192.168.16.XXX:49152 remote 211.XXX.XXX.XXX:34402 to 0
10:28:23 237150691mS CMMap: PCG::MapBChan pcp[546]b0r1 cp_b f545b1e4 other_cp_b f5451ccc type CGTypeDTMF
10:28:23 237150691mS CMMap: a=21.43 b=2.20 M22
10:28:23 237150691mS H323Evt: SESS 1a0 SetOperational local 192.168.16.XXX:49152 remote 211.XXX.XXX.XXX:34402 to 1
10:28:23 237150691mS CMMap: DTMF::playTones g T[447] Tones '2'
10:28:23 237150692mS CMMap: Started playing tones, cp[95]b1r0 (from g G[445])
10:28:23 237150692mS CMMap: cp[95]b1r0 SetOperGroup2 from g G[445] to g G[445] not executed, playing tones in progress 2
10:28:23 237150832mS CMMap: PCG::UnmapBChan pcp[546]b0r1 cp_b f545b1e4 other_cp_b f5451ccc
10:28:23 237150832mS CMMap: a=21.43 b=2.20 M02
10:28:23 237150832mS H323Evt: SESS 1a0 SetOperational local 192.168.16.XXX:49152 remote 211.XXX.XXX.XXX:34402 to 0
10:28:23 237150833mS CMMap: PCG::UnmapBChan pcp[95]b1r0 cp_b f4fb2194 other_cp_b 0
10:28:23 237150833mS CMMap: a=21.43 b=0.0 T0
10:28:23 237150834mS CMMap: a=21.43 b=0.0 DTMF::~DTMF freed CMRTUnknown(f5451de4) resource busy 2, total 45
10:28:23 237150834mS CMMap: PCGS CPReserveCodec for the other end (pcp[546]b0r1) true
10:28:23 237150834mS CMMap: PCG::MapBChan pcp[95]b1r0 cp_b f4fb2194 other_cp_b f545b1e4 type CGTypeSimple
10:28:23 237150835mS CMMap: a=2.20 b=20.3 M12
10:28:23 237150835mS H323Evt: SESS 1a0 SetOperational local 192.168.16.XXX:49152 remote 211.XXX.XXX.XXX:34402 to 1
10:28:23 237150836mS CMMap: a=20.3 b=0.0 PCGS CPReserveCodec (pcp[546]b0r1) true
10:28:23 237150836mS CMMap: a=20.3 b=0.0 PCGS CPReserveCodec (pcp[546]b0r1) true
10:28:23 237150836mS CMMap: PCG::MapBChan pcp[546]b0r1 cp_b f545b1e4 other_cp_b f4fb2194 type CGTypeSimple
10:28:23 237150836mS CMMap: a=20.3 b=2.20 M12
10:28:23 237150840mS CMMap: a=20.3 b=0.0 PCGS CPReserveCodec (pcp[546]b0r1) true
10:28:23 237150840mS CMMap: a=20.3 b=0.0 PCGS CPReserveCodec (pcp[546]b0r1) true
10:28:23 237150840mS CMMap: PCG::MapBChan pcp[546]b0r1 cp_b f545b1e4 other_cp_b f4fb2194 type CGTypeSimple
10:28:23 237150840mS CMMap: a=20.3 b=2.20 M12
10:28:23 237150864mS CMExtnEvt: Kevin: No user activity
10:28:24 237151096mS SIP Rx: UDP 211.XXX.XXX.XXX:5060 -> 192.168.16.XXX:5060
INVITE sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bf02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676416 INVITE
Contact: <sip:211.XXX.XXX.XXX:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 258

v=0
o=BroadWorks 614493471 2 IN IP4 211.XXX.XXX.XXX
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 34402 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=ptime:20
a=maxptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,32,36
a=ptime:20
a=maxptime:20
a=inactive
10:28:24 237151100mS SIP Call Rx: 17
INVITE sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bf02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676416 INVITE
Contact: <sip:211.XXX.XXX.XXX:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 258

v=0
o=BroadWorks 614493471 2 IN IP4 211.XXX.XXX.XXX
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 34402 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=ptime:20
a=maxptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,32,36
a=ptime:20
a=maxptime:20
a=inactive
10:28:24 237151102mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 211.XXX.XXX.XXX:5060 to 211.XXX.XXX.XXX:5060
10:28:24 237151103mS SIP Call Tx: 17
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bf02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676416 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.5.0 build 145
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Content-Length: 0

10:28:24 237151104mS SIP Tx: UDP 192.168.16.XXX:5060 -> 211.XXX.XXX.XXX:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bf02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676416 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.5.0 build 145
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Content-Length: 0

10:28:24 237151106mS Sip: c0a810100000330a 17.13066.0 440 SIPTrunk Endpoint(f160aea8) SetRemoteRTPAddress to 0.0.0.0:34402
10:28:24 237151110mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 211.XXX.XXX.XXX:5060 to 211.XXX.XXX.XXX:5060
10:28:24 237151114mS SIP Call Tx: 17
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bf02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676416 INVITE
Contact: "Tech Support" <sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.5.0 build 145
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Content-Type: application/sdp
Content-Length: 217

v=0
o=UserA 3156239437 878474014 IN IP4 192.168.16.XXX
s=Session SDP
c=IN IP4 192.168.16.XXX
t=0 0
m=audio 49152 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=inactive
10:28:24 237151114mS SIP Tx: UDP 192.168.16.XXX:5060 -> 211.XXX.XXX.XXX:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bf02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676416 INVITE
Contact: "Tech Support" <sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.5.0 build 145
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Content-Type: application/sdp
Content-Length: 217

v=0
o=UserA 3156239437 878474014 IN IP4 192.168.16.XXX
s=Session SDP
c=IN IP4 192.168.16.XXX
t=0 0
m=audio 49152 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=inactive
10:28:24 237151117mS CMMap: a=20.3 b=0.0 PCGS CPReserveCodec (pcp[546]b0r1) true
10:28:24 237151118mS CMMap: PCG::UnmapBChan pcp[546]b0r1 cp_b f545b1e4 other_cp_b f4fb2194
10:28:24 237151118mS CMMap: a=20.3 b=2.20 M02
10:28:24 237151118mS H323Evt: SESS 1a0 SetOperational local 192.168.16.XXX:49152 remote 211.XXX.XXX.XXX:34402 to 0
10:28:24 237151118mS CMMap: PCG::UnmapBChan pcp[95]b1r0 cp_b f4fb2194 other_cp_b 0
10:28:24 237151118mS CMMap: PlatformConnectionAudioSAP::DisconnectRtpFilter rtpRelay ID: 4 Close success
10:28:24 237151118mS CMMap: a=20.3 b=0.0 H1
10:28:24 237151121mS CD: CALL: 0.13062.0 BState=Connected Cut=1 Music=1.0 Aend="Tech(259)" (20.3) Bend="Line 17" [Line 17] (0.0) CalledNum=1800XXXXXX () CallingNum=259 (Tech) Internal=0 Time=11852 AState=Connected
10:28:24 237151221mS SIP Rx: UDP 211.XXX.XXX.XXX:5060 -> 192.168.16.XXX:5060
ACK sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKac7mee300847i1ga0100.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676416 ACK
Contact: <sip:211.XXX.XXX.XXX:5060;transport=udp>
Max-Forwards: 29
Content-Length: 0

10:28:24 237151223mS SIP Call Rx: 17
ACK sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKac7mee300847i1ga0100.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676416 ACK
Contact: <sip:211.XXX.XXX.XXX:5060;transport=udp>
Max-Forwards: 29
Content-Length: 0

10:28:24 237151225mS SIP Rx: UDP 211.XXX.XXX.XXX:5060 -> 192.168.16.XXX:5060
INVITE sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bv02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676417 INVITE
Contact: <sip:211.XXX.XXX.XXX:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp
Max-Forwards: 29
Content-Length: 0

10:28:24 237151228mS SIP Call Rx: 17
INVITE sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bv02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676417 INVITE
Contact: <sip:211.XXX.XXX.XXX:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp
Max-Forwards: 29
Content-Length: 0

10:28:24 237151229mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 211.XXX.XXX.XXX:5060 to 211.XXX.XXX.XXX:5060
10:28:24 237151230mS SIP Call Tx: 17
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bv02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676417 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.5.0 build 145
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Content-Length: 0

10:28:24 237151231mS SIP Tx: UDP 192.168.16.XXX:5060 -> 211.XXX.XXX.XXX:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bv02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676417 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.5.0 build 145
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Content-Length: 0

10:28:24 237151232mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 211.XXX.XXX.XXX:5060 to 211.XXX.XXX.XXX:5060
10:28:24 237151236mS SIP Call Tx: 17
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bv02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676417 INVITE
Contact: "Tech Support" <sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.5.0 build 145
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Content-Type: application/sdp
Content-Length: 264

v=0
o=UserA 3156239437 878474015 IN IP4 192.168.16.XXX
s=Session SDP
c=IN IP4 192.168.16.XXX
t=0 0
m=audio 49152 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=recvonly
10:28:24 237151236mS SIP Tx: UDP 192.168.16.XXX:5060 -> 211.XXX.XXX.XXX:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bv02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676417 INVITE
Contact: "Tech Support" <sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.5.0 build 145
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Content-Type: application/sdp
Content-Length: 264

v=0
o=UserA 3156239437 878474015 IN IP4 192.168.16.XXX
s=Session SDP
c=IN IP4 192.168.16.XXX
t=0 0
m=audio 49152 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=recvonly
10:28:24 237151237mS CMLineRx: v=0
CMFacility
Line: type=SIPLine 17 Call: lid=17 id=13066 in=0
Called[] Type=Default (100) Reason=CMDRdirect Calling[259] Type=Internal Plan=Default
IE CMIERespondingPartyNumber (230)(P:100 S:100 T:0 N:100 R:4) number=1800XXXXXX
IE CMIEDeviceDetail (231) c0a810100000330a LOCALE=ena HW=15 VER=9 class=CMDeviceSIPTrunk type=0 number=17 channel=1 features=0x0 rx_gain=32 tx_gain=32 ep_callid=13066 ipaddr=192.168.16.XXX apps=0 loc=999 em_loc=999 features2=0x0 is_spcall=1
10:28:24 237151241mS CD: CALL: 0.13062.0 BState=Connected Cut=1 Music=1.0 Aend="Tech(259)" (20.3) Bend="Line 17" [Line 17] (0.0) CalledNum=1800XXXXXX () CallingNum=259 (Tech) Internal=0 Time=11973 AState=Connected
10:28:24 237151393mS SIP Rx: UDP 211.XXX.XXX.XXX:5060 -> 192.168.16.XXX:5060
ACK sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKb4rv49003o7ua1geg400.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676417 ACK
Contact: <sip:211.XXX.XXX.XXX:5060;transport=udp>
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 315

v=0
o=BroadWorks 614493471 3 IN IP4 211.XXX.XXX.XXX
s=-
c=IN IP4 211.XXX.XXX.XXX
t=0 0
m=audio 34402 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:30
a=sqn: 0
a=cdsc:1 image udptl t38
a=inactive
10:28:24 237151396mS SIP Call Rx: 17
ACK sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKb4rv49003o7ua1geg400.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676417 ACK
Contact: <sip:211.XXX.XXX.XXX:5060;transport=udp>
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 315

v=0
o=BroadWorks 614493471 3 IN IP4 211.XXX.XXX.XXX
s=-
c=IN IP4 211.XXX.XXX.XXX
t=0 0
m=audio 34402 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:30
a=sqn: 0
a=cdsc:1 image udptl t38
a=inactive

********** Warning: Logging to Screen Stopped **********
 
So it's an internal user that dial the AA ?
Why is it transfered externally?
Why does one number start with 08XXXXXXXX and the other is 1800XXXXXX?

Also you're telling the remote party to contact your internal IP, which would not work unless you have an SBC rewriting that.

10:28:24 237151114mS SIP Tx: UDP 192.168.16.XXX:5060 -> 211.XXX.XXX.XXX:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.XXX.XXX.XXX:5060;branch=z9hG4bKl0kodp0000v8gcl2m500sb4p4bf02.1
From: <sip:1800XXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=1183507842-1485138554175
Call-ID: e2d7026f7409901d59a53554fcaae850
CSeq: 613676416 INVITE
Contact: "Tech Support" <sip:08XXXXXXXX@192.168.16.XXX:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.5.0 build 145
To: "Tech Support" <sip:08XXXXXXXX@sipconnect.XXXXXXXXX.XXX.XX>;tag=735d9686880b6b83
Content-Type: application/sdp
Content-Length: 217

v=0
o=UserA 3156239437 878474014 IN IP4 192.168.16.XXX
s=Session SDP
c=IN IP4 192.168.16.XXX
t=0 0
m=audio 49152 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=inactive

"Trying is the first step to failure..." - Homer
 
Hi ,

We had this issue at a few sites with SIP lines - something to do with the transfer from the IVR to a user or group was confusing the IPO
Avaya sent a critical patch for it "After successive HOLD & UNHOLD at a very fast pace, the talkpath isn''t established"

This critical patch was included in 9.1.9.0.182
 
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