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Hary_CsC
Beginner

Incoming Call Target Session Problem

Hi, guys

 

I've a problem with the session target.
why is the result of the incoming call session target going to CUCM sub1 even though I have directed the dial-peer to go to CUCM pub.
details as follows:

-------------------------------
CUCM pub : 10.74.118.4
CUCM sub1 : 10.74.118.68
-------------------------------

dial-peer voice 1 voip
description *** Outbound LAN Side Dial-Peer (INCOMING CUCM Publisher) ***
preference 1
destination-pattern 4....
session protocol sipv2
session target ipv4:10.74.118.4
voice-class codec 1
voice-class sip bind control source-interface Loopback0
voice-class sip bind media source-interface Loopback0
dtmf-relay rtp-nte sip-kpml h245-alphanumeric h245-signal
!
dial-peer voice 2 voip
description *** Outbound LAN Side Dial-Peer (INCOMING CUCM Subcriber) ***
preference 2
destination-pattern 4....
session protocol sipv2
session target ipv4:10.74.118.68
voice-class codec 1
voice-class sip bind control source-interface Loopback0
voice-class sip bind media source-interface Loopback0
dtmf-relay rtp-nte sip-kpml h245-alphanumeric h245-signal
!

 

* Attached results  #show call active voice compact (session target points to CUCM sub1).

* CUCMG has also made cucm pub the primary.

 

20 REPLIES 20

Based on your shared information I would recommend you to make these changes.

  1. Remove your current dial peers.
  2. Replace with this configuration.

 

voice class uri CUCM sip
 host ipv4:10.74.118.4 ;CPE Publisher
 host ipv4:10.74.118.68 ;CPE Subscriber
;add as many line as there are CPE nodes in the CM cluster
!
voice class uri PSTN sip
 host ipv4:10.18.19.57
!
voice class e164-pattern-map 1
 description E164 Pattern Map for called number to CUCM
  e164 4....
!
voice class e164-pattern-map 2000
 description E164 Pattern Map for called number to PSTN
  e164 9T
!
voice class server-group 1
 ipv4 10.74.118.4 preference 1
 ipv4 10.74.118.68 preference 2
 description Inbound calls from PSTN to CUCM
!
voice class server-group 2000
 ipv4 10.18.19.57 preference 1
;add as many line as there are needed for ITSP service
 description Outbound calls to PSTN ITSP provider
!
voice class sip-options-keepalive 1
 description Used for Server Group SIP OPTIONS PING
!
dial-peer voice 1000 voip
 description Outbound calls from CUCM
 voice-class codec 1
 voice-class sip bind control source-interface Loopback0
 voice-class sip bind media source-interface Loopback0
 session protocol sipv2
 incoming uri via CUCM
 dtmf-relay rtp-nte sip-kpml
 no vad
!
dial-peer voice 1010 voip
 description Inbound calls to CUCM
 session protocol sipv2
 session server-group 1
 destination e164-pattern-map 1
 voice-class codec 1
 voice-class sip options-keepalive profile 1 
 voice-class sip bind control source-interface Loopback0
 voice-class sip bind media source-interface Loopback0
 dtmf-relay rtp-nte sip-kpml
 no vad
!
dial-peer voice 100 voip
 description Inbound calls from PSTN
 translation-profile incoming INCOMING
 session protocol sipv2
 incoming uri via PSTN
 voice-class codec 1
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 110 voip
 description Outbound calls to PSTN
 translation-profile outgoing OUTGOING
 session protocol sipv2
 session server-group 2000
 destination e164-pattern-map 2000
 voice-class codec 1 
 voice-class sip options-keepalive profile 1 ;if ITSP provider has other requirement for SIP options ping create another profile to match their requirements
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 voice-class sip audio forced
 dtmf-relay rtp-nte
 no vad

 Usually I would also have this as a starting point for any SBC (Cube) setups.

 

[A.A.A.A] = LAN Interface IP Address (Voice Vlan)  
[B.B.B.B] = Assigned IP address from ITSP (Outside Interface) 
[C.C.C.C] = ITSP SIP SBC IP Address
[D.D.D.D] = ITSP CPE IP address 
Note: Please note that for some telco, SIP SBC is same as CPE IP address

voice class sip-profiles 10 ;modify as needed for ITSP service requirement
 request ANY sip-header From modify "A.A.A.A" "B.B.B.B" ;to hide internal IP from SIP FROM field
 request ANY sip-header From modify "From:(.*)(<sip:.*@.*>)" "From: \2" 
 response ANY sip-header From modify "From:(.*)(<sip:.*@.*>)" "From: \2" 
 request ANY sip-header Remote-Party-ID modify "Remote-Party-ID:(.*)(<sip:.*@.*>)" "Remote-Party-ID: \2" 
 response ANY sip-header Remote-Party-ID modify "Remote-Party-ID:(.*)(<sip:.*@.*>)" "Remote-Party-ID: \2" 
 request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)(<sip:.*@.*>)" "P-Asserted-Identity: \2" 
 response ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)(<sip:.*@.*>)" "P-Asserted-Identity: \2" 
!
dial-peer voice 110 voip
 description Outbound calls to PSTN
 voice-class sip profiles 10

 



Response Signature


Hary_CsC
Beginner

Hi @Roger Kallberg 

 

Thank you for the reply awesome for the solution.

I'll try the configuration. 

 

Hi @Scott Leport 

Any suggestions for RTP to CUCM PUB?

 

Thank

Hi,


Difficult to say for sure, but a couple of suggestions:

 

1. MTPs configured in your MRG, same MRG added to MRGL on CUCM applied to SIP trunk between CUCM and CUBE

2. SCCP CCM order on your CUBE

 

That's working on the assumption that MTP is applied and at play here.

Do you possibly have the "Media Termination Point Required" check box set on your SIP trunk configuration element in CM?

image.png

If so please try to uncheck it and reset the trunk in CM.

For this it would also be advisable to verify this setting on the SIP profile in CM before you reset the trunk.

image.png



Response Signature


Hi @Roger Kallberg 

 

When i was uncheck Media Termination Point Required, phone call can't.

 

Thank

The fact that you need to use a MTP for your calls to succeed would indicate you’ll likely have something that is not all correctly configured for the connection with you SIP telephony service provider. Normally you should not need to have a MTP involved in the call for it to succeed.

Can you please share the entire configuration, as it stands now after incorporating the suggested alterations, of your voice gateway and also screenshots of the complete configuration for the SIP trunk for your voice gateway in CM, including the complete SIP profile. Also please post screenshots of your Device Pool, MRGL, MRG and MTP configuration for the involved elements for the call.

Likely there is something around DTMF signaling method that could be the issue for why a MTP is needed, but without the entire picture of the above it's not all that easy to say for sure. One thing that you can check is that you have this set on the trunk.

image.png



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