Hi,
we have a CM 6.0 with direct SIP Trunk to another SIP-PBX (such as Asterisk).
the junction works quite well. Users logged on SIP-PBX using ISDN trunks on the CM for inbound and outbound.
I want to know if the CM is the ability to manipulate the Information Element in the ISDN Setup Message for an outgoing call that comes from the SIP-PBX. In fact, for these calls the Calling Party Number is not correct because the Type of Number is "Unknow" and Numberig Plan is "Private". The digits are OK.
{isdn}<---> [CM] <---SIP-TRK---> [SIP-PBX] <-->Phone
These are the settings.
---------------------------------------------------------------------------------------------
TRUNK GROUP
Group Number: 20 Group Type: sip CDR Reports: r
Group Name: SIP_PBX COR: 96 TN: 1 TAC: 1920
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: public-ntwrk Auth Code? n
Member Assignment Method: auto
Signaling Group: 20
Number of Members: 12
Group Type: sip TRUNK PARAMETERS
Unicode Name: auto
Redirect On OPTIM Failure: 5000
SCCAN? n Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 600
Disconnect Supervision - In? y Out? y
XOIP Treatment: auto Delay Call Setup When Accessed Via IGAR? n
TRUNK FEATURES
ACA Assignment? n Measured: none
Maintenance Tests? y
Numbering Format: public
UUI Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Modify Tandem Calling Number: no
Show ANSWERED BY on Display? y
PROTOCOL VARIATIONS
Mark Users as Phone? n
Prepend '+' to Calling Number? n
Send Transferring Party Information? y
Network Call Redirection? y
Send Diversion Header? n
Support Request History? y
Telephone Event Payload Type:
Convert 180 to 183 for Early Media? n
Always Use re-INVITE for Display Updates? y
Identity for Calling Party Display: P-Asserted-Identity
Enable Q-SIP? n
-------------------------------------------------------------------------------------------------------------------
SIGNALING GROUP
Group Number: 20 Group Type: sip
IMS Enabled? y Transport Method: tcp
Q-SIP? n SIP Enabled LSP? n
IP Video? n Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: Others
Near-end Node Name: procr Far-end Node Name: SIP_PBX
Near-end Listen Port: 5060 Far-end Listen Port: 5060
Far-end Network Region: 5
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? n
Alternate Route Timer(sec): 6
-------------------------------------------------------------------------------------------
Thank you.
we have a CM 6.0 with direct SIP Trunk to another SIP-PBX (such as Asterisk).
the junction works quite well. Users logged on SIP-PBX using ISDN trunks on the CM for inbound and outbound.
I want to know if the CM is the ability to manipulate the Information Element in the ISDN Setup Message for an outgoing call that comes from the SIP-PBX. In fact, for these calls the Calling Party Number is not correct because the Type of Number is "Unknow" and Numberig Plan is "Private". The digits are OK.
{isdn}<---> [CM] <---SIP-TRK---> [SIP-PBX] <-->Phone
These are the settings.
---------------------------------------------------------------------------------------------
TRUNK GROUP
Group Number: 20 Group Type: sip CDR Reports: r
Group Name: SIP_PBX COR: 96 TN: 1 TAC: 1920
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: public-ntwrk Auth Code? n
Member Assignment Method: auto
Signaling Group: 20
Number of Members: 12
Group Type: sip TRUNK PARAMETERS
Unicode Name: auto
Redirect On OPTIM Failure: 5000
SCCAN? n Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 600
Disconnect Supervision - In? y Out? y
XOIP Treatment: auto Delay Call Setup When Accessed Via IGAR? n
TRUNK FEATURES
ACA Assignment? n Measured: none
Maintenance Tests? y
Numbering Format: public
UUI Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Modify Tandem Calling Number: no
Show ANSWERED BY on Display? y
PROTOCOL VARIATIONS
Mark Users as Phone? n
Prepend '+' to Calling Number? n
Send Transferring Party Information? y
Network Call Redirection? y
Send Diversion Header? n
Support Request History? y
Telephone Event Payload Type:
Convert 180 to 183 for Early Media? n
Always Use re-INVITE for Display Updates? y
Identity for Calling Party Display: P-Asserted-Identity
Enable Q-SIP? n
-------------------------------------------------------------------------------------------------------------------
SIGNALING GROUP
Group Number: 20 Group Type: sip
IMS Enabled? y Transport Method: tcp
Q-SIP? n SIP Enabled LSP? n
IP Video? n Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: Others
Near-end Node Name: procr Far-end Node Name: SIP_PBX
Near-end Listen Port: 5060 Far-end Listen Port: 5060
Far-end Network Region: 5
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? n
Alternate Route Timer(sec): 6
-------------------------------------------------------------------------------------------
Thank you.
