Утро доброе.
Подскажите, с чем может быть связана ошибка в трейсе при входящем вызове на sip-транк?

SIP>SIP/2.0 488 Not Acceptable Here (Transport Not Secure)

Настройки сигналинга и транка:

Voice System name: SIGNALING GROUP

Group Number: 29 Group Type: sip
IMS Enabled? n Transport Method: tcp
Q-SIP? n
IP Video? n Enforce SIPS URI for SRTP? n
Peer Detection Enabled? y Peer Server: Others
Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers? n
Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers? y
Alert Incoming SIP Crisis Calls? n
Near-end Node Name: procr Far-end Node Name: Call-Center
Near-end Listen Port: 5060 Far-end Listen Port: 5060
Far-end Network Region:

Far-end Domain:
Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? n
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? y
Alternate Route Timer(sec): 6

LIMIT SIGNALING GROUP USAGE

Enable on the main Processor(s)? y

Enable on Survivable Processors (ESS and LSP): all



SIP-Trunk
Voice System name: TRUNK GROUP

Group Number: 29 Group Type: sip CDR Reports: y
Group Name: Call-Center COR: 3 TN: 1 TAC: 0629
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: tie Auth Code? n
Member Assignment Method: auto
Signaling Group: 29
Number of Members: 30

Group Type: sip

TRUNK PARAMETERS

Unicode Name: yes

Redirect On OPTIM Failure: 5000

SCCAN? n Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 900

Disconnect Supervision - In? y Out? y


XOIP Treatment: auto Delay Call Setup When Accessed Via IGAR? n

Caller ID for Service Link Call to H.323 1xC: original-calling-number

TRUNK FEATURES
ACA Assignment? n Measured: none
Maintenance Tests? y

Numbering Format: public
UUI Treatment: service-provider

Replace Restricted Numbers? y
Replace Unavailable Numbers? y


Modify Tandem Calling Number: no

Show ANSWERED BY on Display? n

PROTOCOL VARIATIONS

Mark Users as Phone? y
Prepend '+' to Calling/Alerting/Diverting/Connected Number? n
Send Transferring Party Information? n
Network Call Redirection? n

Send Diversion Header? n
Support Request History? y
Telephone Event Payload Type: 96
Convert 180 to 183 for Early Media? n
Always Use re-INVITE for Display Updates? n
Identity for Calling Party Display: P-Asserted-Identity
Block Sending Calling Party Location in INVITE? n
Accept Redirect to Blank User Destination? n
Enable Q-SIP? n

Interworking of ISDN Clearing with In-Band Tones: keep-channel-active


Сам трейс, вызов не поступает при звонке из колл-центра на внутренние номера в организации (Avaya)