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Hary_CsC
Beginner

Voice Gateway - Reject Call Doesn't Get Cut Off

Hi Guys
 
I just implemented a voice gateway, for outgoing calls there is no problem.
But when I try to reject it doesn't get cut off but still "beep beep beep" as if it's still ringing.
 
(I tried to call out to my personal cellphone then I refused (Reject call), but the result was still "beep beep beep")
 
Here is the debug result.
 
Thank you
1 ACCEPTED SOLUTION

Accepted Solutions

Try with this configuration instead of yours

voice service voip
 ip address trusted list
  ipv4 10.40.1.81
  ipv4 10.204.93.81
  ipv4 10.204.93.82
  ipv4 10.254.93.81
 rtcp keepalive
 address-hiding
 mode border-element
 media statistics
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711ulaw
 modem passthrough nse codec g711alaw
 trace
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  session refresh
  error-passthru
  no update-callerid
  midcall-signaling passthru media-change
  early-offer forced


voice class uri CUCM sip
 host ipv4:10.204.93.81
 host ipv4:10.204.93.82
 host ipv4:10.254.93.81
!
voice class uri PSTN sip
 host 10.40.1.81


voice class server-group 1
 ipv4 10.204.93.81 preference 1
 ipv4 10.204.93.82 preference 2
 ipv4 10.254.93.81 preference 3
 description Inbound calls to CUCM
!
voice class server-group 2000
 ipv4 10.40.1.81 preference 1
 description Outbound calls to PSTN
!
voice class sip-options-keepalive 1
 description Used for Server Group SIP OPTIONS PING


voice class e164-pattern-map 1
 description E164 Pattern Map for called number to CUCM
  e164 72.....
!
voice class e164-pattern-map 2000
 description E164 Pattern Map for called number to PSTN
  e164 9T


no dial-peer voice 1
no dial-peer voice 2
no dial-peer voice 3
no dial-peer voice 101
no dial-peer voice 102
no dial-peer voice 103
no dial-peer voice 104
no dial-peer voice 105
no dial-peer voice 106

dial-peer voice 1000 voip
 description  *** Incoming Dial-Peer from CUCM  ***
 session protocol sipv2
 incoming uri via CUCM
 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
 fax-relay sg3-to-g3
 no vad
!
dial-peer voice 1010 voip
 description  *** Inbound calls from PSTN to CUCM (Server group 1)  ***
 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
 fax-relay sg3-to-g3
 no vad
!
dial-peer voice 100 voip
 description  *** Incoming Dial Peer from SIP GRATIKA  ***
 redirect ip2ip
 session protocol sipv2
 incoming uri via PSTN
 voice-class codec 1  
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 no vad
!
dial-peer voice 110 voip
 description *** Outbound Dial Peer to SIP GRATIKA ***
 translation-profile outgoing GRATIKA_OUTGOING
 session protocol sipv2
 session server-group 2000
 destination e164-pattern-map 2000
 voice-class codec 1  
 voice-class sip options-keepalive profile 1
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 voice-class sip audio forced
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 no vad
 

dspfarm profile 1 transcode
no max session
 no codec g729abr8
maximum sessions 75

voice class codec 1
 no codec preference 4 g729br8


Response Signature


View solution in original post

8 REPLIES 8
Roger Kallberg
VIP Mentor

Likely your telco isn’t happy with the response that you send, as seen in your ladder diagram, that btw is not really the same as output of debug. If we’re truly to be able to give you help with this your best option is to get the whole output of debug ccsip message and attach it as a file.

A question, for what reason have you masked out the private IP addresses in the ladder diagram?



Response Signature


Hai @Roger Kallberg 

Have a Nice day.

Sorry it's a photo taken from my colleague.

I've a debug sample from another VG whose reject sound is sent to the extension (IP Phone), attached.

 

Could this problem be from VG or CUCM ? or even Provider ?

 

Thank You

It is likely related to configuration in your gateway (SBC). For what reason do you have multiple dial peers that almost all are the same for the outbound path to your service provider? The output of debug ccsip message is needed to actually see what’s going on in the SBC for the call.



Response Signature


Hi @Roger Kallberg 

 

If what you say is a problem with SBC, it means the provider (ISP) right?

Multiple dial-peer for me to be able to call out to random numbers. for example:

021-29978801

021-29978802

021-29978803

021-29978804

021-29978805

021-29978806

021-2997880x

021-299788xx

 

Thank You

I am asking since all dial peers from 101 to 106 has this

destination-pattern 9T
session target ipv4:10.40.1.81

With this how would that do this?


@Hary_CsC wrote:

 

Multiple dial-peer for me to be able to call out to random numbers. for example:

021-29978801

021-29978802

021-29978803

021-29978804

021-29978805

021-29978806

021-2997880x

021-299788xx


Any of these dial peers will let you call to a number that starts with a 9 and send it to the same session target 10.40.1.81.

 

No with SBC I mean your gateway that runs the Cube functionality. It acts as a Session Border Controller. More specifically I mean that there is a problem with the configuration in this specific gateway in relation with what this specific service provider expects to get for a rejected call. In your two ladder diagrams you send different cause codes for the rejected calls.

For the working as you say you send this.

image.png

For the non working you send this

image.png



Response Signature


Hai @Roger Kallberg  

 

Multiple dial-peers to issue random numbers on outgoing calls

 

voice translation-rule 1
rule 1 /^7\(......$\)/ /02129978801/
!
voice translation-rule 2
rule 1 /^9\(.*\)/ /\1/
!
voice translation-rule 3
rule 1 /^7\(......$\)/ /02129978802/
!
voice translation-rule 4
rule 1 /^7\(......$\)/ /02129978803/
!
voice translation-rule 5
rule 1 /^7\(......$\)/ /02129978804/
!
voice translation-rule 6
rule 1 /^7\(......$\)/ /02129978805/
!
voice translation-rule 7
rule 1 /^7\(......$\)/ /02129978806/

 

---------------------------------------------------------------------------------

Yes, it's different from other voice gateways because the providers are different.

Why is the VG JYP-RTR-RCR-VG-01 code 480 Temporarily ..... (101 INVITE) ? because I think it was sent from the provider.

 

In the configuration you provided there is incoming call, but I don't use it.

Thanks in advance maybe i will try it first.

 

BR

HS

I think that you might need to read up on how call handling works in a voice gateway. There is always an inbound dial peer used and an outbound. These can be the same, but cater to the two call legs that each call comprise of, one inbound call leg and one outbound.

Have a look at this document for details on this.

https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html

If you would not have masked your configuration in the attached file it would have made it easier to understand your use case with sending different calling numbers per dial peer.



Response Signature


Try with this configuration instead of yours

voice service voip
 ip address trusted list
  ipv4 10.40.1.81
  ipv4 10.204.93.81
  ipv4 10.204.93.82
  ipv4 10.254.93.81
 rtcp keepalive
 address-hiding
 mode border-element
 media statistics
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711ulaw
 modem passthrough nse codec g711alaw
 trace
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  session refresh
  error-passthru
  no update-callerid
  midcall-signaling passthru media-change
  early-offer forced


voice class uri CUCM sip
 host ipv4:10.204.93.81
 host ipv4:10.204.93.82
 host ipv4:10.254.93.81
!
voice class uri PSTN sip
 host 10.40.1.81


voice class server-group 1
 ipv4 10.204.93.81 preference 1
 ipv4 10.204.93.82 preference 2
 ipv4 10.254.93.81 preference 3
 description Inbound calls to CUCM
!
voice class server-group 2000
 ipv4 10.40.1.81 preference 1
 description Outbound calls to PSTN
!
voice class sip-options-keepalive 1
 description Used for Server Group SIP OPTIONS PING


voice class e164-pattern-map 1
 description E164 Pattern Map for called number to CUCM
  e164 72.....
!
voice class e164-pattern-map 2000
 description E164 Pattern Map for called number to PSTN
  e164 9T


no dial-peer voice 1
no dial-peer voice 2
no dial-peer voice 3
no dial-peer voice 101
no dial-peer voice 102
no dial-peer voice 103
no dial-peer voice 104
no dial-peer voice 105
no dial-peer voice 106

dial-peer voice 1000 voip
 description  *** Incoming Dial-Peer from CUCM  ***
 session protocol sipv2
 incoming uri via CUCM
 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
 fax-relay sg3-to-g3
 no vad
!
dial-peer voice 1010 voip
 description  *** Inbound calls from PSTN to CUCM (Server group 1)  ***
 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
 fax-relay sg3-to-g3
 no vad
!
dial-peer voice 100 voip
 description  *** Incoming Dial Peer from SIP GRATIKA  ***
 redirect ip2ip
 session protocol sipv2
 incoming uri via PSTN
 voice-class codec 1  
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 no vad
!
dial-peer voice 110 voip
 description *** Outbound Dial Peer to SIP GRATIKA ***
 translation-profile outgoing GRATIKA_OUTGOING
 session protocol sipv2
 session server-group 2000
 destination e164-pattern-map 2000
 voice-class codec 1  
 voice-class sip options-keepalive profile 1
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 voice-class sip audio forced
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 no vad
 

dspfarm profile 1 transcode
no max session
 no codec g729abr8
maximum sessions 75

voice class codec 1
 no codec preference 4 g729br8


Response Signature


View solution in original post

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