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| Note |
|---|
Dues to a limitation where the gateway configured in sipxecs always uses port 5060, OSBC can not be used for FENT and SIP Trunking simultaneously. |
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1.a Registering a NAT'd UA to sipxecs through OSBC with UpperReg(without and with domain rewriting)
In OSBC:
Setting OSBC with UpperReg Mode:
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- Setup your phone
Uid: 200 Pw: 200
Domain: sipx.example.com <-- this domain should resolve to the IP address of your SipXecs instance
Proxy: osbc.example.com <-- tjis domain should resolve to the IP address of your OSBC instance
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1.b Calls between UAs behind NAT
Callflow: UA1 ? OSBC ? SIPX ? OSBC ? UA2
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- Setup your phones
Uid: 200 Pw: 200
Domain: sipx.example.com <-- this domain should resolve to the IP address of your SipXecs instance
Proxy: osbc.example.com <-- this domain should resolve to the IP address of your OSBC instance
Uid: 201 Pw: 201
Domain: sipx.example.com <-- this domain should resolve to the IP address of your SipXecs instance
Proxy: osbc.example.com <-- this domain should resolve to the IP address of your OSBC instance
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1.c Calls to PSTN, UAs behind NAT
Callflow: UA1 ? OSBC ? SIPX ? OSBC _-> PSTN Gateway
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- Setup your phones
Uid: 200 Pw: 200
Domain: sipx.example.com <-- your SipXecs instance
Proxy: osbc.example.com <-- your OSBC instance
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1.d Calls From PSTN, UAs behind NAT
Callflow: PSTN Caller ? OSBC ? SIPX ? OSBC ? UA1
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| Note |
|---|
Dues to a limitation where the gateway configured in sipxecs always uses port 5060, OSBC can not be used for FENT and SIP Trunking simultaneously. |
...
2.a Calls to PSTN, UAs and SipXecs on a Local Network
Callflow: UA1 ? SIPX ? OSBC _-> PSTN Gateway
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- Setup your phones
Uid: 200 Pw: 200
Domain: sipx.example.com <-- your SipXecs instance
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2.b Calls from PSTN, UAs and SipXecs on a Local Network
Callflow: PSTN Caller ? OSBC ? SIPX ? UA1
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| Note |
|---|
Dues to a limitation where the gateway configured in sipxecs always uses port 5060, OSBC can not be used for FENT and SIP Trunking simultaneously. |
...
3.a Registering OSBC to the SIP Trunk*
In OSBC:
Setting OSBC to Register to a Sip Trunk Provider:
- Go to OSBC Sip Trunk Config
- Paste the XML for the config (sample below)
Code Block <root> <siptrunk trunk-name="my.sipprovider.com" route-set="sip:sip.sipprovider.com" sip-domain="sip.sipprovider.com" expires="10"> <trunk-accounts> <account user-name="xxxxxxx" auth-user-name="xxxxxx" auth-password="xxxxxx" inbound-route="sip:100@sipx.example.com" expires="3600"/> </trunk-accounts> <transient-accounts> <account user-name="xxxxx" auth-user-name="xxxx" auth-password="xxxx" inbound-route="sip:100@sipx.example.com" expires="3600"/> </transient-accounts> </siptrunk> </root>
- Press Update
- To check if registration is successful, Go to Sip-Trunk Registration Status
sip:xxxxxxx@sip.sipprovider.com sip:xxxxxx@xxx.xxx.xx.xx:5066 00773236-91be-dd11-8be9-e96beb558260@sip.sipprovider.com SIP/2.0 200 OK
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3.b Routing of inbound calls from a trunk provider to the sipxecs autoattendant
Callflow: Caller ? SipTrunk Provider ? OSBC ? SipX
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- Setup your phones
Uid: 200 Pw: 200
Domain: sipx.example.com <-- your SipXecs instance
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3.c Routing of outbound calls from a trunk provider
Call Flow: UA1 ? SIPX ? OSBC _-> Sip Trunk Provider
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