Troubleshooting Guide for SIP Caller
Welcome to the SIP Caller troubleshooting guide. This page is designed to help you identify and resolve common issues you may encounter while using the SIP Caller platform. Whether you're experiencing problems with extension registration, campaign calls not being made, or issues with call connectivity, this guide will walk you through the basic steps to diagnose and correct the problem.
Please review each section carefully to find guidance on specific issues.
Extension Registration Issues
If your SIP extension is not registering with the PBX, there are a few common causes to investigate. Expand each issue below for more details:
Wrong Credentials — 401 Unauthorized
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.
Invalid FQDN — Host 'invalid.example.com' could not be resolved
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.
Firewall Blocking Traffic — Timeout
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.
SIP Caller Does Not Make Calls for Campaign
If a campaign is not generating calls, several factors may be responsible. Expand each item below to check potential causes:
Check “Active” State
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".
Check Date, Time, and Time Zone
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.
Check Holidays
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.
Check That the Campaign Has Numbers
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.
In Free or Demo Accounts, Check the Calls Per Day Limit
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.
Check if Other Campaigns Are Running and Consuming Simultaneous Calls
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.
In Predictive Dialer Mode, Verify That the Proxy Is Installed and the Queue Has Available Agents
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.
Calls Generated by SIP Caller Do Not Go Through
If SIP Caller attempts to place calls but they fail immediately, the problem is usually related to PBX or SIP trunk configuration. Expand the items below to investigate:
No Outbound Rule Configured — IVR with announcement
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.
SIP Trunk Has No Balance — 403 Forbidden
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.
Country Blocked — 403 Forbidden
or IVR with announcement
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.
Invalid Destination Number — 404 Not Found
or IVR with announcement
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.
No Route to Destination — 404 Not Found
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.
Problems with Established Calls
If calls are being connected but no audio is heard or the call is dropped shortly after starting, there are a few likely causes. Expand the sections below for details:
Secure RTP (SRTP) Mismatch
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:
SIP Caller | 3CX | Result |
---|
Secure RTP Enabled | SRTP Enabled | ✅ Audio works |
Secure RTP Enabled | SRTP Disabled | ❌ No audio |
Secure RTP Enabled | SRTP Enforced | ✅ Audio works |
Secure RTP Disabled | SRTP Enabled | ✅ Audio works |
Secure RTP Disabled | SRTP Disabled | ✅ Audio works |
Secure RTP Disabled | SRTP Enforced | ❌ No audio |
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.
Failed Transfers
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.
DTMF Digits not Detected
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:
- 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.
- Confirm the DTMF transport method - In Wireshark, under Telephony > RTP > RTP Streams, verify that DTMF tones are not using RFC 2833.
- 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.
- Verify PBX configuration - Ensure your PBX is set to support and prefer RFC 2833 for DTMF signaling.
- Test again - Make another call, capture the RTP, and confirm that DTMF tones are now being sent as RTP events.
Final Notes
If after following these steps your issue is not resolved, feel free to contact our support team for further assistance.