Versions Compared

Old Version 6

changes.mady.by.user Former user

Saved on

compared with

New Version Current

changes.mady.by.user Former user

Saved on

Key

  • This line was added.
  • This line was removed.
  • Formatting was changed.
Comment: Migrated to Confluence 5.3

...

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.

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:

...

  1. 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

1.b Calls between UAs behind NAT

Callflow: UA1 ? OSBC ? SIPX ? OSBC ? UA2

...

  1. 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

1.c Calls to PSTN, UAs behind NAT

Callflow: UA1 ? OSBC ? SIPX ? OSBC _-> PSTN Gateway

...

  1. Setup your phones
    Uid: 200 Pw: 200
    Domain: sipx.example.com <-- your SipXecs instance
    Proxy: osbc.example.com <-- your OSBC instance

1.d Calls From PSTN, UAs behind NAT

Callflow: PSTN Caller ? OSBC ? SIPX ? OSBC ? UA1

...

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

...

  1. Setup your phones
    Uid: 200 Pw: 200
    Domain: sipx.example.com <-- your SipXecs instance

2.b Calls from PSTN, UAs and SipXecs on a Local Network

Callflow: PSTN Caller ? OSBC ? SIPX ? UA1

...

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:

  1. Go to OSBC Sip Trunk Config
  2. 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>
  3. Press Update
  4. 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

3.b Routing of inbound calls from a trunk provider to the sipxecs autoattendant

Callflow: Caller ? SipTrunk Provider ? OSBC ? SipX

...

  1. Setup your phones
    Uid: 200 Pw: 200
    Domain: sipx.example.com <-- your SipXecs instance

3.c Routing of outbound calls from a trunk provider

Call Flow: UA1 ? SIPX ? OSBC _-> Sip Trunk Provider

...