Explanation:
This usually indicates that the username, authentication ID or password configured for the extension is incorrect.

How to fix:

• Double-check the SIP username, authentication ID and password in the SIP Caller configuration.
• Make sure the extension is active and not blocked or disabled on the PBX.
• If you recently changed the password in the PBX, update it in SIP Caller as well.

Explanation:
This indicates that the domain or IP address specified for the PBX server is incorrect or unreachable.

How to fix:

• Verify that the domain name (FQDN) is spelled correctly.
• Make sure the domain points to the correct public IP address of your PBX.
• Check that the DNS service can resolve the domain.

Explanation:
Network-level firewalls or PBX settings may be blocking SIP or RTP traffic.

How to fix:

• Make sure SIP (usually UDP or TCP port 5060) is open on the PBX firewall.
• Ensure any cloud or hosting provider firewalls allow traffic from SIP Caller's IP address.
• Check your PBX logs for rejected registration attempts or blocked IPs.
• When using 3CX Phone System, ensure that the Block remote non tunnel connection option is disabled for the extension used by SIP Caller.

Explanation:
A campaign state must be set to "Active" to start placing calls.

How to fix:

• Go to the Campaigns section in the SIP Caller console.
• Make sure the campaign is set to "Active" and not "Paused", "Finished" or "Canceled".

Explanation:
Campaigns only run during their scheduled time window and respect the configured time zone.

How to fix:

• Confirm that the current date and time fall within the campaign's schedule.
• Ensure the correct time zone is selected in the campaign settings.

Explanation:
If a campaign has a holiday group set, the current day scheduling might be affected.

How to fix:

• Check the "Holiday Group" for the campaign.
• Remove or adjust holidays as needed to allow the campaign to run.

Explanation:
Campaigns with no phone numbers cannot make calls.

How to fix:

• Go to the campaign’s list of numbers.
• Ensure at least one valid number is loaded and not marked as completed or failed.

Explanation:
Free accounts are limited to 100 calls per day, and Demo accounts are limited to 500 calls per day.

How to fix:

• Monitor your daily call volume on the dashboard.
• Wait until the next day or upgrade your account to remove the limit.

Explanation:
Simultaneous call capacity is shared across all running campaigns.

How to fix:

• Pause or stop other active campaigns to free up call capacity.
• Upgrade your plan if you need more simultaneous calls.

Explanation:
Predictive campaigns require a SIP Caller Proxy to be installed and a queue with available agents.

How to fix:

• Confirm the SIP Caller Proxy is running and correctly connected to your PBX.
• Ensure agents are logged into the queue and available to take calls.

Explanation:
If there is no outbound rule that matches the dialed number, the PBX may redirect the call to an IVR or reject it.

How to fix:

• Check the outbound rules in your PBX to ensure they match the dialing format used by SIP Caller (e.g., with or without country code, prefix).
• Add or adjust rules so the numbers in your campaign can be routed to the SIP trunk.

Explanation:
Some SIP providers will reject calls with a 403 Forbidden response if your account balance is insufficient.

How to fix:

• Log into your SIP trunk provider's portal and check your account balance.
• Recharge or top up your account to allow outbound calls.

Explanation:
Your PBX or SIP trunk may be configured to block international or specific country destinations.

How to fix:

• Review outbound call permissions in the PBX and the SIP trunk settings.
• Enable international or country-specific dialing if needed.

Explanation:
The number format may be invalid or the number might not exist.

How to fix:

• Make sure the numbers in your campaign are valid and correctly formatted for your PBX and SIP trunk.
• Use E.164 formatting if required by your provider.

Explanation:
Your SIP trunk provider might not have routing available for the dialed number.

How to fix:

• Contact your SIP trunk provider to verify coverage for the dialed destination.
• Consider using a backup trunk or alternate provider for those destinations.

Explanation:
A mismatch in Secure RTP settings between SIP Caller and your PBX (e.g. 3CX) can result in no audio, even if the call connects successfully.

Typical Scenarios:

How to fix:

• Ensure that SRTP is enabled or disabled on both ends consistently.
• If using 3CX with enforced SRTP, enable Secure RTP in SIP Caller settings.
• If 3CX has SRTP disabled, disable SRTP in SIP Caller as well.

Explanation:
In predictive dialing or queue-based scenarios, the PBX may reject the call transfer if there are no available agents in the target queue.

How to fix:

• Check the queue configuration in your PBX.
• Ensure that the queue has agents logged in and available to receive calls.

Explanation:
When DTMF tones are transmitted as part of the audio stream (in-band), audio compression can distort the high-frequency signals, making them unreliable for detection. To ensure accurate DTMF recognition, tones must be sent using the RTP payload method defined in RFC 2833.

How to fix:

1. Capture the RTP stream using Wireshark - On the PBX side, run a packet capture during a test call and inspect the RTP packets:
   • Look for DTMF tones being transmitted as audio (in-band).
   • If you don’t see RTP events (Payload Type 101 or similar), tones are likely being sent in-band.
2. Confirm the DTMF transport method - In Wireshark, under Telephony > RTP > RTP Streams, verify that DTMF tones are not using RFC 2833.
3. Contact your SIP Trunk provider - Share your findings and ask them to configure DTMF to be sent using RFC 2833 (RTP Payload Events) instead of in-band audio.
4. Verify PBX configuration - Ensure your PBX is set to support and prefer RFC 2833 for DTMF signaling.
5. Test again - Make another call, capture the RTP, and confirm that DTMF tones are now being sent as RTP events.