06-24-2015 05:40 PM - edited 03-13-2019 09:02 PM
Hi Guys
I am having issue in incoming call in SRST
Jun 24 12:10:23.944: ISDN Se0/1/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0029
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA1839A
Preferred, Channel 26
Calling Party Number i = 0x2181, '2653312732'
Plan:ISDN, Type:National
Called Party Number i = 0x81, '905'
Plan:ISDN, Type:Unknown
*Jun 24 12:10:23.948: ISDN Se0/1/0:15 Q931: Received SETUP callref = 0x8029 callID = 0x000C switch = primary-net5 interface = User
*Jun 24 12:10:23.956: ISDN Se0/1/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8029
Channel ID i = 0xA9839A
Exclusive, Channel 26
*Jun 24 12:10:23.956: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x8029
Cause i = 0x8081 - Unallocated/unassigned number
*Jun 24 12:10:24.112: ISDN Se0/1/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x0029
Cause i = 0x8081 - Unallocated/unassigned number
*Jun 24 12:10:24.112: ISDN Se0/1/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x8029
*Jun 24 12:10:42.320: ISDN Se0/1/0:15 Q931: RX <- SETUP pd = 8 callref = 0x002A
ANy help would be appreciated
Solved! Go to Solution.
06-24-2015 09:58 PM
Neither your debugs are complete nor your config.
Anyways it seems to me your problem is since Telco is delivering 3 digits and your extensions are 4 digit, and this is happening only in SRST mode, it appears to me you don't have digit manipulation setup to convert 3 digits to 4 digit extension 79XX.
Either you can have separate dial-peer for incoming numbers and apply this on dialpeer or you can apply this to call-manager-fall to take effect only during SRST. As during normal operations CUCM/MGCP takes control of call routing.
Try the below:
voice translation-rule 100
rule 1 /\(.*\)/ /7\1/
voice translation-profile SRST-IN
translate called 100
call-manager-fall
translation-profile incoming SRST-IN
Let me know how you go with this. Also make sure your phone is correctly registered in SRST mode before you make the test.
-Terry
Please rate all helpful posts
06-24-2015 05:53 PM
Unallocated number means misconfiguration somewhere, do you have incoming/outgoing dial-peers for 901?
Can you attach your config along the post and also make another test call and post the output of below:
1) debug isdn q931
2) debug voip ccapi input
3) debug voip dialpeer inout
-Terry
06-24-2015 08:25 PM
HI Terry/Jamie
Thanks for your replies
I have 4 digit dialing pattern 7XXX, but ISP provides 3 digits also attaching the config of router
debug voip ccapi inout
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3E6E92E8
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 7914
Called Number : 7903
Source IP Address (Sig ): 10.191.33.20
Destn SIP Req Addr:Port : 10.191.22.62:5060
Destn SIP Resp Addr:Port : 10.191.22.62:5060
Destination Name : 10.191.22.62
*Jun 24 12:14:30.892: //145916/6061F497BB0A/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): 10.191.33.20
Source IP Port (Media): 17240
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Jun 24 12:14:30.892: //145916/6061F497BB0A/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 487
*Jun 24 12:14:30.896: //145915/6061F497BB0A/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3E6DC6A8
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 7914
Called Number : 7903
Source IP Address (Sig ): 10.191.33.20
Destn SIP Req Addr:Port : 10.191.22.64:5060
Destn SIP Resp Addr:Port : 10.191.22.64:5060
Destination Name : 10.191.22.64
*Jun 24 12:14:30.896: //145915/6061F497BB0A/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.191.33.20
Source IP Port (Media): 17238
Destn IP Address (Media): 10.191.22.64
Destn IP Port (Media): 16392
Orig Destn IP Address:Port (Media): [ - ]:0
*Jun 24 12:14:30.896: //145915/6061F497BB0A/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 487
06-24-2015 09:58 PM
Neither your debugs are complete nor your config.
Anyways it seems to me your problem is since Telco is delivering 3 digits and your extensions are 4 digit, and this is happening only in SRST mode, it appears to me you don't have digit manipulation setup to convert 3 digits to 4 digit extension 79XX.
Either you can have separate dial-peer for incoming numbers and apply this on dialpeer or you can apply this to call-manager-fall to take effect only during SRST. As during normal operations CUCM/MGCP takes control of call routing.
Try the below:
voice translation-rule 100
rule 1 /\(.*\)/ /7\1/
voice translation-profile SRST-IN
translate called 100
call-manager-fall
translation-profile incoming SRST-IN
Let me know how you go with this. Also make sure your phone is correctly registered in SRST mode before you make the test.
-Terry
Please rate all helpful posts
06-24-2015 11:27 PM
Hi Terry
I will check and confirm to you
06-25-2015 05:58 AM
Hi Terry
Thanks for all your help :)
06-24-2015 07:57 PM
If this goes to CUCM check all your inbound call routing from the GW or trunk, significant digits, inbound CSS, etc.
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