Below are instructions for sending own CLI in Asterisk (FreePBX). They differ from typical Asterisk setup somewhat. See the generic Asterisk/FreePBX/Elastix setup guide if you are registering your new account with DID Logic to your Asterisk server.
This setup guide is only intended for verified business customers. We do not offer passthrough CLI for the majority of SIP accounts. Under normal circumstances, your caller ID should be changed by using your didlogic.com account controls (DID numbers in your "caller ID" dropdown list). Certain business customers may be eligible for custom CLI option if they absolutely require sending their existing numbers as CLI, or passing through the Caller ID information of the forwarded call. Before you can transmit your own custom CLI to the PSTN networks, your organization will need to provide proof of physical location (your electricity, water, natural gas or building society bill that arrives via postal mail - no PDFs, no screenshots, and no electronic bills are accepted).
1 Change "type=friend" to "type=peer"; remove "fromuser"
Valid registry string and active registration are still required to send calls: 12345:*****@sip.didlogic.net
- sip.didlogic.net resolves to closest SIP proxy. Business customers will be provided access to specific SIP gateways based on their account preferences and other requirements (like, for example, Amsterdam, Frankfurt, Paris, Johannesburg, New York, Singapore or Sydney)
- there is support for GSM and G729, yet we absolutely insist you use G711u or G722 codecs. Admittedly there are some places where bandwidth is expensive, however, low bitrate codecs usage always results in SIP provider voice quiality that we shall call "quite unimpressive" for censorship reasons.
- your 5-digit SIP username and password are different from your didlogic.com web login credentials. Set these up in the "SIP" tab of your didlogic.com account. IMPORTANT: 3 incorrect SIP auth attempts will ban your IP address for 3 hours.
2 Outbound route setup
In this particular example, we're defining the trunk with the "Allow
any CID" option, and specify the CLI to be sent in the route.
However, you can also define CID in the trunk and set the outbound
route to obey trunk CID (and/or ignore extension), or allow the route
to pass extension CID on a per-call basis.
Important: dialing format is E164. Dialing with 00 or 011 in front will not work. You need to send the dialed number using the international format, with country code, area code and number (1 for NANPA countries). Dialing US/Canada requires a "1" in front.
442012345678 – United Kingdom 12125551212 – USA
19055551212 – Canada 4915151234567 – Germany
011442012345678 or 00442012345678 or 02012345678 – this is NOT how you dial UK. 2125551212 or 9055551212 – this is NOT how you call US/Canada, you must dial with "1" in front.
CLI sent as 5 digits, or your DID number only Set to "No caller ID" and check debug first!
- Please set the CLI for the SIP sub-account you wish to use with dynamic CLI to "No Caller ID" prior to requesting the feature.
- Turn on debug by issuing the "sip set debug ip enter_proxy_ip_here" when attempting to send calls, and examine the output. If different from below example, your gateway is not transmitting the CLI in the appropriate filed:
Syntax example when sending +12125551212 as CLI via the London proxy:
From: "anything" <sip:12125551212@your_IP_addr_here>;tag=as028d2be5
FAQ If you still can't make calls - checklist to go through prior to contacting support
- 603 declined on outbound is caused by lack of active registration. SIP registration is currently required to send calls.
- Dial the full international number: 1-212-555-1212 (US), 44-208-500-0000 (London). Dialing 212-555-1212 won't work. Do not dial with 0 or 00 or 011 in front. This is by far the most frequent reason for outgoing calls to fail.
- Incoming calls problem: issue the "sip set debug ip sip.didlogic.net" command and review incoming traffic from us. 99.8% of such issues are caused by wrong context or other incorrect route setup. Each Asterisk installation is unique. You may have various contexts enabled that interact in your own customized way; please look at your syntax first.
- If in doubt, stop all registrations, comment out the trunk config, download X-lite and attempt placing calls. This will tell you right away that the problem is somewhere in your local Asterisk configuration.
Reminder: be very sure you are using valid SIP username and password to register and send calls. Our system will ban your IP address if you attempt incorrect credentials. You will not be able to browse any pages at didlogic.com, or login to your account to make changes for at least 3 hours, and the ban will be renewed for another 3 hours if your Asterisk fails to authenticate again. That cycle may potentially go on forever. Please ensure you are using the 5-digit username and your own secure password created at the "SIP" tab of your didlogic.com account.