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Beginner

Incoming Calls issue from ITSP to VGW via SIP Trunk

Hi Everyone, 

I have an issue with Inbound calls from ITSP that has been connected to my cisco VGW through SIP trunk, I got the following disconnect cause:

 

Disconnect Cause (CC) : 57
Disconnect Cause (SIP) : 403

appreciate you support to figure out this issue, 
Attached are ccsip debug files
Thanks and Regards, 

1 ACCEPTED SOLUTION

Accepted Solutions
Highlighted

 

Your inbound  dial-peers are

 

dial-peer voice 2000 voip
description #TO_Y Inside_01#
answer-address [1-8]...$
destination-pattern 21
session protocol sipv2
session target ipv4:10.11.200.20
dtmf-relay sip-info sip-kpml sip-notify rtp-nte
codec g711alaw
!
dial-peer voice 2001 voip
description #TO_Y Inside_02#
answer-address [1-8]...$
destination-pattern 21
session protocol sipv2
session target ipv4:10.11.200.21
dtmf-relay sip-info sip-kpml sip-notify rtp-nte
codec g711alaw

Rules for Matching Inbound Dial Peers

The five configurable dial peer attributes are:

  • Incoming called number--A string representing the called number or DNIS. It is configured by using the incoming called-numberdial-peer voice configuration command in POTS or multimedia mail over IP (MMoIP) dial peers.

     

  • Answer address--A string representing the calling number or ANI. It is configured by using the answer-address dial-peer voice configuration command in POTS or VoIP dial peers and is used only for inbound calls from the IP network.

     

  • Destination pattern--A string representing the calling number or ANI. It is configured by using the destination-pattern dial-peer voice configuration command in POTS or voice-network dial peers.

     

  • Application--A string representing the predefined application that you wish to enable on the dial peer. It is configured by using the applicationdial-peer voice configuration command on inbound POTS dial peers.

     

  • Port--The voice port through which calls to this dial peer are placed.

Your inbound call information

 

The Call Setup Information is:

Call Control Block (CCB) : 0x0x7F62F9E8AEE8

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : +96477xxxxxxx4

Called Number            : 78xxxxxxx8

There is nothing to match with your dial-peer configuration.

 

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View solution in original post

13 REPLIES 13
Highlighted

Hi, 

Do you have the correct codec defined in the dial peers? 

is there any possibility that you can share debug sip message for the inbound calls? 

Regards, 

Highlighted
VIP Advocate

Can you share your VG configuration and debug ccsip messages

 

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Highlighted
Beginner

Thank you for replying, herewith debug ccsip msg and Run config files.

Thanks in advance, 

Highlighted

Hi, 

Could you please clarify the device name with IP .214 and .213? 

  • Received a call from Zain 
  • The call is sending to .213 
  • you are receiving 403 Forbidden from.213 
  • similarly, it repeats on .214
  1. Do you have any entity with 7835416308 in 213 and/214? 
  2. Does the .213/.214 accept the G711alaw codec?

 

Could you please let us know the call flow? 

Zain (ITSP) --> CUBE (.250) --> .213/.214 (Is it CUCM? ) 

 

Regards,  

Shalid

Highlighted

Hi Shalid, 

I'm using this VGW t connect ITSP from a side (SIP servers 172.21.175.212, 213 and 214), from other side it connects to internal servers used for contact center (10.11.200.20, and 21 don't think it shown in debug).
So, for inbound call flow:

ITSP (SIP Signal 172.21.175.212-214, RTP Traffic 10.x.x.100) --->> Voice Gateway (Dial-peer 1000 > dest dpg 2) --->> CC Servers (10.11.200.20-21)

for now, I got different cause code 
Disconnect Cause (SIP) : 480 CC: 31

Disconnect Cause (SIP) : 487 CC: 102


Thanks 

Highlighted

Thanks for the details. 

 

as @Nithin Eluvathingal since you don't have a dial-peer for the destination number to the call manager, it is taking the route back to ITSP via dial-peer 1013 and 1014. 

could you please modify your 2001 dial peer for testing as below and make a call. 

 

dial-peer voice 2001 voip
description #TO_CUCM#
destination-pattern 78........$ (or any of DID) --> this is based on your first call sample. 
session protocol sipv2
session target ipv4:10.11.200.21
dtmf-relay sip-info sip-kpml sip-notify rtp-nte
codec g711alaw

no vad

 

Either use TP to translate the DID to an extension or choose significant digit = 4 in your SIP trunk configuration page depends on your dial plan. 

 

Regards 

 

 

 

Highlighted

 

Your inbound  dial-peers are

 

dial-peer voice 2000 voip
description #TO_Y Inside_01#
answer-address [1-8]...$
destination-pattern 21
session protocol sipv2
session target ipv4:10.11.200.20
dtmf-relay sip-info sip-kpml sip-notify rtp-nte
codec g711alaw
!
dial-peer voice 2001 voip
description #TO_Y Inside_02#
answer-address [1-8]...$
destination-pattern 21
session protocol sipv2
session target ipv4:10.11.200.21
dtmf-relay sip-info sip-kpml sip-notify rtp-nte
codec g711alaw

Rules for Matching Inbound Dial Peers

The five configurable dial peer attributes are:

  • Incoming called number--A string representing the called number or DNIS. It is configured by using the incoming called-numberdial-peer voice configuration command in POTS or multimedia mail over IP (MMoIP) dial peers.

     

  • Answer address--A string representing the calling number or ANI. It is configured by using the answer-address dial-peer voice configuration command in POTS or VoIP dial peers and is used only for inbound calls from the IP network.

     

  • Destination pattern--A string representing the calling number or ANI. It is configured by using the destination-pattern dial-peer voice configuration command in POTS or voice-network dial peers.

     

  • Application--A string representing the predefined application that you wish to enable on the dial peer. It is configured by using the applicationdial-peer voice configuration command on inbound POTS dial peers.

     

  • Port--The voice port through which calls to this dial peer are placed.

Your inbound call information

 

The Call Setup Information is:

Call Control Block (CCB) : 0x0x7F62F9E8AEE8

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : +96477xxxxxxx4

Called Number            : 78xxxxxxx8

There is nothing to match with your dial-peer configuration.

 

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=>>>If you find this response useful, please mark it as "HELPFUL"<<<=

View solution in original post

Highlighted

Hi Nithin, 

Dial-peer voice 2000 and 2001 should be forward inbound calls to CC servers (DN consist of 4 digits XXXX) which is the answer-address, n the other hand Dial-peer 1000 should be receive the calls from ITSP and forward to Dial-peers 2000&2001 (DPG 2).

ITSP SIP Trunk-->DP 1000  >> DP 2000&2001 (DN XXXX) -->> CC servers SIP Trunk

Please correct me if wrong, 

Thanks 

Highlighted

your called number is 78xxxxxxx8 and your dial peer is for 4 digit.

Use  translation to convert 78xxxxxxx8  to 4 digit .

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Highlighted

The OP is trying to use DPG to route the calls, then the legacy match part of the dial peers should be irrelevant.

Please rate all useful posts
Highlighted

I hope your internal numbers are 4 digit, if yes translate 78xxxxxxx8 to a 4 digit and sent it to CC or do it from CC if possible.

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Highlighted

exactly, my internal number are 4 digit and I did configured translation-rule but still cannot receive the call.
Also, I did register an IP Phone to VGW with extension 1000 so the call flow will be: 
ITSP-->> VGW-->>IP Phone

So, for inbound calls I've to configure a dial-peer.. can you please share the correct configuration of this dial-peer?? and the right expression for translation-rule?

Thanks a lot, 

Highlighted

Try below

 

voice translation-rule 1
rule 1 /^78....\([1-8]...$\)/ /\1/

 

voice translation-profile Incoming
translate called 1

 

dial-peer voice 2000 voip

translation-profile incoming Incoming

 


dial-peer voice 2001 voip
translation-profile incoming Incoming


Testing Voice translation Rule
jlrvg#test voice translation-rule 1 7845122154
Matched with rule 1
Original number: 7845122154 Translated number: 2154
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none

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