11-11-2021 11:25 AM
I currently have a Cisco 4321 router that is setup with a single PRI and a single SIP trunk. We are adding a different SIP provider to this router. I believe I need to use the "voice class tenant" config for the second SIP provider. The current SIP configuration on the router uses credentials configured under "sip-ua". The new SIP provider does not require credentials. Also the current SIP provider is using a single IP address where as the new provider has 4 addresses. So my questions are, (1) using the Voice class Tenant configuration can I set it up to use multiple IP addresses? If so what does that look like. (2) if I don't have credentials for the new SIP provider is having credentials listed under "sip-ua" going to impact the voice class tenant configuration? (3) Does applying a voice class tenant to a dial-peer take precedence over the commands listed in the "sip" section of the "voice service voip" profile? and (4) what other configurations are required under the voice class tenant profile? Hopefully this makes sense. Thank for the help.
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11-12-2021 06:38 AM
The day got a little crazier than expected, so I never got the time to look at your specifics in your configuration, sorry about that. However the below would be an example on how a register config could looks like if you have it in a tenant configuration instead of in sip-ua.
voice class tenant 2000 registrar dns:<service provider DNS name> expires 3600 credentials username <user name for connection> password 7 <password for connection> realm <service provider realm name> authentication username <user name for connection> password 7 <password for connection> no remote-party-id timers dns registrar-cache 95 sip-server dns:<service provider DNS name> connection-reuse audio forced bind control source-interface GigabitEthernet0/0/1 bind media source-interface GigabitEthernet0/0/1 no pass-thru content custom-sdp sip-profiles 10 outbound-proxy dns:<service provider DNS name for proxy> reuse early-offer forced ! dial-peer voice 100 voip description Inbound calls from PSTN translation-profile incoming PSTN-IN session protocol sipv2 incoming uri via PSTN voice-class codec 2000 voice-class sip tenant 2000 dtmf-relay rtp-nte no vad ! dial-peer voice 110 voip description Outbound calls to PSTN translation-profile outgoing PSTN-OUT session protocol sipv2 session server-group 2000 destination e164-pattern-map 2000 voice-class codec 2000 voice-class sip tenant 2000 voice-class sip options-keepalive profile 2000 dtmf-relay rtp-nte no vad
11-11-2021 12:51 PM - edited 11-11-2021 12:51 PM
Once in front of a computer tomorrow I’ll give you a more detailed answer. It would be of great help if you could share your current configuration, redacted for any sensitive information.
In short you would move the current registration information from sip-ua into a tenant configuration as that’s a per tenant configuration, whereas sip-ua is global.
As a start you might have a look at this document. It has many valuable pieces of information.
11-11-2021 12:56 PM
This is from the document linked below, which is a chapter in the "CUBE Book":
Iftenantsareconfiguredunderdial-peer,thenconfigurationsareappliedinthefollowingorderofpreference.
• Dial-peer configuration
• Tenant configuration
• Global configuration
For SIP trunk registration, the voice class tenant <tag> command is not associated with any dial-peer
configuration.AlloutgoingregistrationsaretriggeredtotheRegistrarswhencredentialsareconfiguredunder
voice class tenant <tag>.
Router# show run | sec tenant
Voice class tenant 1
registrar 1 ipv4:10.64.86.35:9051 expires 3600
credentials username aaaa password 7 06070E204D realm aaaa.com
outbound-proxy ipv4:10.64.86.35:9057
bind control source-interface GigabitEthernet0/0
Voice class tenant 2
registrar 1 ipv4:9.65.75.45:9052 expires 3600
credentials username bbbb password 7 110B1B0715 realm bbbb.com
outbound-proxy ipv4:10.64.86.40:9040
bind control source-interface GigabitEthernet0/1
11-11-2021 01:48 PM
@Roger Kallberg, Thank you for replying. I've attached the current config. I will read through the document you posted in the meantime. Thank you
11-12-2021 06:38 AM
The day got a little crazier than expected, so I never got the time to look at your specifics in your configuration, sorry about that. However the below would be an example on how a register config could looks like if you have it in a tenant configuration instead of in sip-ua.
voice class tenant 2000 registrar dns:<service provider DNS name> expires 3600 credentials username <user name for connection> password 7 <password for connection> realm <service provider realm name> authentication username <user name for connection> password 7 <password for connection> no remote-party-id timers dns registrar-cache 95 sip-server dns:<service provider DNS name> connection-reuse audio forced bind control source-interface GigabitEthernet0/0/1 bind media source-interface GigabitEthernet0/0/1 no pass-thru content custom-sdp sip-profiles 10 outbound-proxy dns:<service provider DNS name for proxy> reuse early-offer forced ! dial-peer voice 100 voip description Inbound calls from PSTN translation-profile incoming PSTN-IN session protocol sipv2 incoming uri via PSTN voice-class codec 2000 voice-class sip tenant 2000 dtmf-relay rtp-nte no vad ! dial-peer voice 110 voip description Outbound calls to PSTN translation-profile outgoing PSTN-OUT session protocol sipv2 session server-group 2000 destination e164-pattern-map 2000 voice-class codec 2000 voice-class sip tenant 2000 voice-class sip options-keepalive profile 2000 dtmf-relay rtp-nte no vad
11-12-2021 09:14 AM
@Roger KallbergNo worries, I know how it goes. Thank you for taking the time to help me. I assume once I set up the registered tenant config I will need to remove the "sip-ua" registration info of will that not matter anymore because the router will be looking at the tenant config? Also, for the new SIP trunk without registration requirement I would set it up the same minus the registration information? Sorry for all the questions I haven't setup a config like this before and don't want to take down the current SIP trunk. Oh, hopefully, one last thing. Does the "voice service voip" section need to be modified one the tenants are setup? Thank again for the help.
11-12-2021 09:41 AM
Hi,
No worries, we’re all been new to various parts at one time.
Thats correct, you would remove the current registration information from the sip-ua. Also correct about the none registration SIP connection, set that up with a tenant without registration configuration. No there are nothing specific that you need to configure under voice service voip.
11-12-2021 09:58 AM
@Roger Kallberg Thank you again. Your help is much appreciated.!
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