How do I setup my SIP trunk for outbound calling? Asterisk

We authenticate PPW SIP traffic by source IP address on our platform. After you add your gateway IP addresses to your allowed list through the TCXC portal, you will need to establish a SIP trunk with our proxy servers (, port 5060).

On open source applications such as Asterisk (Open Source PBX) you can setup your SIP trunk as below. 

Please note that no SIP registration is required against our platform.


There are several GUI interfaces for Asterisk that simplify installation of Asterisk. These interfaces allow administrators to view, edit and change most aspects of Asterisk via a Web interface.

 If you are not an advanced user of Asterisk, we highly recommend using one of the following GUI interfaces:

On other types of switch platforms, we recommend that you configure your switch as per your vendor guidelines (see Appendix). 

Please note that we do not provide direct technical support for customer PPW switch platforms. 

2) Which CODECs do you support?
We support G711, G729 as well as G723. Generally speaking, we recommend that our 

customers offer G711 as well as G729 in their initial SIP INVITE to us. 

3) How do I setup my dialing plan for outbound calling?
Below is the recommended dialing plan that should be setup on your switch: Aggregator / PPW Dialing Plan: 

Default Routing with dialing prefix: 

• NANP format (1 + 10 digits and 011 + CC+ . . .) • E164 (Must be with leading +)
• 00 + CC + . . .

Carrier and dialing prefix: 

• 1234 + NANP format (1 + 10 digits and 011 + CC + . . .) • 1234 + CC +. . . (i.e. E164 with no leading +)
• 1234 + 00 + CC + . . .

Please note that 10 digit dialing for North American countries is NOT supported. For more information on the NANP (North American Numbering Plan), please visit For more information on E164, please visit 

            4) Which format should I use when setting up Caller ID in my switch? 

We recommend using the PAID (P-Asserted-Identity) option for CLI as per RFC 3325 ( Our platform also supports RPID (Remote-Party-ID). Please note that Caller ID is guaranteed over our Platinum routes (prefix 99901) only. Over other divisions, CLI is best effort.

5) Which DTMF settings should I set on my switch? 

We recommend that you enable RFC 2833 payload type 101 for DTMF. Please note that DTMF is supported over our CLI routes (E.g Airtel Gold) only. 


6) Does my switch need to support SIP re-INVITEs?

We have a 15 minute SIP session timer set on all incoming SIP traffic, which means
that we will send a SIP re-INVITE to your switch 15 minutes into the call. Please make sure that you are allowing SIP re-INVITEs on your switch. If you are not sure how to enable SIP re-INVITEs on your specific switch, please reach out to your vendor.

7) How can I capture a SIP trace on my switch?
When setting up a new SIP trunk with a provider or troubleshooting call failures, it's important

to be able to capture a signaling trace of an outbound call.

In Asterisk, you can activate SIP debugging via the Asterisk CLI using the SIP set debug commands:

SIP set debug peer on

Turns on SIP debugging globally showing all SIP traffic to and from the Asterisk gateway

SIP set debug IP

Allows you to debug only to and from a particular IP address

SIP set debug off

Turns off all SIP debugging

If you don't want to enable debugging on your switch, you can use a network protocol analyzer such as Wireshark to capture the SIP and media traffic on your calls. To learn more about Wireshark, please visit This site includes step-by-step videos on how to setup Wireshark on your network.

8) Which protocol do you support for fax transmissions?
Generally speaking, we support T.38 protocol as well as G711 pass-through for fax

transmissions. The majority of our carriers prefer T.38 protocol.

9) I have added an IP address to my portal allowed list but none of my calls are completing. What should I do??

Please note that after adding a new IP address to your allowed list via the port, it can take up to 10 minutes for the changes to go into effect on our platform. Please retest after 10 minutes. If your calls are still not completing, please open a ticket with our support group.


Support links to various switch vendor platforms:

Asterisk - Support Forum

3CX - Support Forum

Avaya - Support Forum

Elastix - Support Forum

Trixbox - Support Forum

Grandstream - Support Forum

Cisco - Support Forums

FreePBX - Support Forum

Talkswitch - Support Forum

AudioCodes - Support

Voipswitch - Support

IPsmarx - Support

Huawei - Support

Sonus Networks - Support



For more details please contact our support helpdesk:



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