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13443
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5
Helpful
6
Replies

Unallocated/unassigned number

yogesh bhalerao
Beginner
Beginner

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

 

 

1 Accepted Solution

Accepted Solutions

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

 

View solution in original post

6 Replies 6

Terry Cheema
VIP Alumni
VIP Alumni

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

 

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

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

 

Hi Terry

I will check and confirm to you 

Hi Terry

 

Thanks for all your help :)

Jaime Valencia
Cisco Employee
Cisco Employee

If this goes to CUCM check all your inbound call routing from the GW or trunk, significant digits, inbound CSS, etc.

HTH

java

if this helps, please rate
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