This article explains the configuration steps necessary to allow SIP calls in an IP.PBX(1) –
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Ribbon SBC 1000/2000 – IP.PBX(2) topology as illustrated in the diagram below. For the purpose of this article, it is assumed that the IP.PBX - Border Elements - are configured to send/receive SIP calls:
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SBC 1000/2000 Configuration
Configuration Overview
The following configuration steps must be accomplished in the order presented as shown below.
- Tone Table — You may use the default Tone Table or create a custom table.
- Media Profiles — In this configuration, G.711A/Mu-law profiles are configured and used.
- Media List — You may use the default Media List or create a custom list.
- SIP Server Tables — Add an entry for border elements (i.e., the IP_PBX's used in this example).
- SIP Profile — You may use the default SIP Profile or create a custom profile.
- Transformation Table — Create a transformation table to be used in the relevant Call Routing Table(s) for Called/Calling Number transformations.
- Call Routing Table — Create two Call Routing Tables (one for ISDN and another for SIP) to be used in the relevant Signaling Groups.
- Signaling Groups — Create two SIP Signaling Groups (one for each border element).
- Call Routing Table Entries — Create two Call Routing Table Entries to route the calls.
Tone Table
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Tone Tables.
Add a Tone Profile Table (named SIPtrunking: Tone Table)
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1 | Add Tone Profile Table |
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- Name them SIPtrunking: G.711 A-Law Voice and SIPtrunking: G.711 Mu-Law Voice.
- Select the G.711 A-Law Codec
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1 | Add New Voice Codec Profiles |
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- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Media > Media List.
- Add a new Media List.
- Specify the Media Profiles created in the previous section in the Media Profiles List text box.
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- The rest of the fields are left with
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1 | Add New Media List |
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SIP Server Tables
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to SIP > SIP Server Tables.
- Add two new SIP Server Table entries (one for each border element).
- Name them SIPtrunking: IP-PBX-1 and SIPtrunking: IP-PBX-2.
Configure the new entries from the previous step
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1 | Add New SIP Server Table Entries |
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SIP Profile
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to SIP > SIP Profiles.
- Add a new SIP Profile
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- (i.e, SIPTrunking: SIP Profile).
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1 | SIP Profile Table |
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- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Transformation.
- Add two new entries and name them SIPtrunking: Translation for IP-PBX-1 ext. and SIPtrunking: Translation for IP-PBX-2 ext..
Configure the entries created in the previous step
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1 | Transformation Table |
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Call Routing Table
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Call Routing Table.
- Add two new Call Routing Tables, and name them SIPtrunking: to IP-PBX-1 and SIPtrunking: to IP-PBX-2.
Signaling Groups
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Signaling Groups
- Add two new SIP Signaling Groups
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SIP Signaling Group #1 |
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SIP Signaling Group #2
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- . Configure with the configuration added in previous steps (i.e., Media List, SIP Profile, SIP Server Table, etc.)
Call Routing Table Entries
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Call Routing Table > (relevant entry).
- Configure each of Call Routing entries that you created earlier in this article as shown below.
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Call Routing Entry #1 for Table #1 |
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Call Routing Entry #1 for Table #2
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Call verification
Once the above configuration steps have been completed, a SIP call from Border Element 1 (IP-PBX-1) to Border Element 2 (IP-PBX-2) using
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SIP Trunking will have the following call flow: Panel |
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