Hi there. I am using CUCM 8.0(3a) and have a SIP Trunk to a Voice gateway. Currentky, I can call a mobile number direct from the logged in DN (using combination of Device CSS and DN CSS). When i run a "debug ccsip calls" on SIP Trunk I gte follloiwng results:-
GBLON-IX31-VG01#
Dec 22 10:59:34.224 GMT: ISDN Se0/1/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x1 0x1, Calling num 442071442580
Dec 22 10:59:34.228 GMT: ISDN Se0/1/0:15 Q931: Sending SETUP callref = 0x00F0 callID = 0x8071 switch = primary-net5 interface = User
Dec 22 10:59:34.228 GMT: ISDN Se0/1/0:15 Q931: TX -> SETUP pd = 8 callref = 0x00F0
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Calling Party Number i = 0x1181, '442071442580'
Plan:ISDN, Type:International
Called Party Number i = 0xA1, '07808908220'
Plan:ISDN, Type:National
Dec 22 10:59:34.420 GMT: ISDN Se0/1/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x80F0
Channel ID i = 0xA98381
Exclusive, Channel 1
Dec 22 10:59:39.472 GMT: ISDN Se0/1/0:15 Q931: RX <- ALERTING pd = 8 callref = 0x80F0
Dec 22 10:59:43.792 GMT: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x00F0
Cause i = 0x8090 - Normal call clearing
Dec 22 10:59:43.796 GMT: //6833/8E0F34800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x6B00D198
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : +442071442580
Called Number : +447808908220
Source IP Address (Sig ): 10.239.255.10
Destn SIP Req Addr:Port : 10.192.20.5:5060
Destn SIP Resp Addr:Port : 10.192.20.5:51625
Destination Name : 10.192.20.5
Dec 22 10:59:43.796 GMT: //6833/8E0F34800000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 10.239.255.10
Source IP Port (Media): 19166
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
Dec 22 10:59:43.796 GMT: //6833/8E0F34800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 487
Dec 22 10:59:44.072 GMT: ISDN Se0/1/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x80F0
Dec 22 10:59:44.072 GMT: ISDN Se0/1/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x00F0
GBLON-IX31-VG01#
Notice Calling Number andd Called Number are populated.
When i try to call forard to same number and call my extension, I gte the folloiwng results:-
GBLON-IX31-VG01#
Dec 22 11:01:49.816 GMT: //6835/DE8692000000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x6AFFA260
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number :
Called Number :
Source IP Address (Sig ): 10.239.255.10
Destn SIP Req Addr:Port : 10.192.20.5:0
Destn SIP Resp Addr:Port : 10.192.20.5:51625
Destination Name : 10.192.20.5
Dec 22 11:01:49.816 GMT: //6835/DE8692000000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 100
Disconnect Cause (SIP) : 400
Dec 22 11:01:49.832 GMT: //6836/DE8692000000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x6AFD9ED8
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number :
Called Number :
Source IP Address (Sig ): 10.239.255.10
Destn SIP Req Addr:Port : 10.192.20.5:0
Destn SIP Resp Addr:Port : 10.192.20.5:51625
Destination Name : 10.192.20.5
Dec 22 11:01:49.832 GMT: //6836/DE8692000000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 100
Disconnect Cause (SIP) : 400
GBLON-IX31-VG01#
Notice this time that there is no Calling or Called Number details being passed to SIP Trunk (Gateway).
Now both of these calls (Direct and Call Forward) are using same Translation Patterns and Route Patterns and going to same SIP Trunk. The onkly difference is the Call Forward option seems to be stripping the Calling and Called Numbers and as a result failing to match any dial-peers on Voice gateway.
All heklp is huigekly appreciated.
