09-23-2021 01:01 PM
Solved! Go to Solution.
09-24-2021 05:10 AM
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
09-23-2021 10:21 PM - edited 09-23-2021 10:24 PM
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?
09-24-2021 12:47 AM
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
09-24-2021 02:58 AM
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.
09-24-2021 05:25 AM
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
09-24-2021 07:05 AM - edited 09-24-2021 07:14 AM
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.
For the non working you send this
09-24-2021 09:56 AM
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
09-24-2021 11:11 AM - edited 09-24-2021 11:16 AM
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.
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.
09-24-2021 05:10 AM
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
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