Switching between MGCP and SIP
I have come across a situation where i have to create a sip trunk to External Contact center in the existing MGCP gateway.I have created basic SIP configurations and dial peers to CX. However when a make a incoming test call it isn’t hitting any dial peer not even dial peer 0 . After few hours of troubleshooting i missed the basic thing i have to perform before switching between MGCP and SIP which is causing my incoming calls to fail in the gateway with error message Unallocated/unassigned number
011063: May 25 14:07:41.824: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x3796
Sending Complete
Bearer Capability i = 0x9090A3
Standard = CCITT
Transfer Capability = 3.1kHz Audio
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA18381
Preferred, Channel 1
Progress Ind i = 0x8A81 – Call not end-to-end ISDN, may have in-band info
Calling Party Number i = 0x1183, ‘XXXXXXX’
Plan:ISDN, Type:International
Called Party Number i = 0xA1, ‘XXXXXX’
Plan:ISDN, Type:Unknown
*Jun 24 12:10:23.956: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x8029
Cause i = 0x8081 – Unallocated/unassigned number
So here is the configuration done did to make the inbound calls working
*********FOR SIP CONFIGURATION***************
voice-port 0/0/0:15
shut
int Serial0/0/0:15
shut
no isdn bind-l3 ccm-manager
controller T1 0/0/0
shut
no pri-group timeslots 1-24 service mgcp
pri-group timeslots 1-24
no shut
voice-port 0/0/0:15
no shut
int Serial0/0/0:15
no shut
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Hi Aravindan,
I’m a little confused about your post. You mentioned changing from MGCP to SIP, but the configurations are for PRI. Did you just have to remove the PRI configurations for it to work?
Also, under your t1 controller, you have both shut and no shut printed:
controller T1 0/0/0
shut
no pri-group timeslots 1-24 service mgcp
pri-group timeslots 1-24
no shut
Otherwise, great job and keep up the good work!
After further review, it appears you may be using your voice gateway to convert from PRI to SIP back to CUCM… So the previous call flow was:
PSTN > PRI > Voice Gateway > MGCP > CUCM
now it is:
PSTN > PRI > Voice Gateway > SIP > CUCM
Is that correct?
So basically you had to remove the MGCP command from the PRI and the L3 bind from the serial port.