SIP Trunking Troubleshooting 

Issue:  Incorrect email address sent
Resolution: Verify 'Advanced SIP Settings' to ensure that the correct IP address is being sent.Address.

Issue: Codec mismatch or codec resource unavailable.  IPComms only supports G.729 and G.711 codecs
Resolution: Check that you have codecs G.729 or G.711 activated on your PBX.

Issue: Your account has reached its call capacity limit. 
Resolution: Order additional SIP trunk lines for your account or reduce the volume of calls.

Issue: Bad SIP username and/or password
Resolution: Verify that you are using the correct SIP username and password for your particular IPComms SIP trunk.

Issue: Firewall blocking SIP request
Resolution: Verify that you have whitelisted IPComms SIP servers and IP Addresses in your PBX and firewall.

Issue: Incorrect number format
Resolution: Ensure that you are sending your calls with preceding "1".  (e.g. 1-6784601475, not  6784601475) for all outbound calls.

Issue: Jitter is one of the most common VoIP call quality problems.
Resolution: Jitter is the variability over time of the latency across a network.  Speak to your Internet service provider about normalizing network latency.  Network hardware can also be a possible issue.  Check your network router and firewall quality and configuration.  

Issue: Poor Internet Connection
Resolution: Check your Internet quality.  Try running PING tests and network traceroutes to check the quality of your Internet connection.  If you need assistance, you can contact a member of our support team to help you isolate network issues.  In addition, contact your ISP for assistance with network quality issues.

Issue: Your firewall is blocking outbound RTP packets to IPComms
Resolution: Ensure your firewall is configured to allow RTP from your PBX.

Issue: Your PBX is using your LAN IP address rather than your Wan IP address for SDP.
Resolution: Ensure your PBX is setup to use the WAN IP address for SDP.

Issue: Your PBX and our network cannot agree on a common codec.
Resolution: Ensure that your PBX has G.729 and G.711 codecs enabled.

Issue: Your firewall is blocking outbound RTP packets to IPComms
Resolution: Speaker phones are very prone to echo. Try asking the other party to turn the volume down on their phone. If they are using a speaker phone see if lowering the speaker volume or picking up the handset eliminated the echo.

Issue: Your PBX does not have the IPComms proxy IP addresses allowed (whitelisted) in your firewall.
Resolution: Confirm that the SIP URI is correct in your PBX origination settings.
Resolution: Be sure that IPComms IP addresses and ports are whitelisted in your PBX (if applicable).
Resolution:Be sure that IPComms IP addresses and ports are whitelisted in your Firewall (if applicable).

Issue: The SIP ID and password aren’t correct.
Resolution: Check the SIP ID and password on the phone.

Issue: There are NAT issues on the firewall or in the Asterisk configuration file. 
Resolution: Your personal firewall on your network might have NAT (network address translation) enabled, or it might be blocking the ports that need to be open for VoIP to communicate to the outside world. There are two types of traffic that need to be forwarded: SIP signaling and RTP media. The default port for UDP based SIP signaling is port 5060. The RTP media traffic (the actual audio stream) uses a range of 10000-20000.

Issue:  The SIP line isn’t registered.
Resolution: Verify your SIP account registration information in your provisioning letter sent in by IPComms upon sign-up.  Ensure it is the same in your device's SIP account settings.

Issue: Jitter is one of the most common VoIP call quality problems.
Resolution: Jitter is the variability over time of the latency across a network.  Speak to your Internet service provider about normalizing network latency.  Network hardware can also be a possible issue.  Check your network router and firewall quality and configuration.  

Issue: Poor Internet Connection
Resolution: Check your Internet quality.  Try running PING tests and network traceroutes to check the quality of your Internet connection.  If you need assistance, you can contact a member of our support team to help you isolate network issues.  In addition, contact your ISP for assistance with network quality issues.

Issue: Your firewall is blocking outbound RTP packets to IPComms
Resolution: Ensure your firewall is configured to allow RTP from your PBX.

Issue: Your PBX is using your LAN IP address rather than your Wan IP address for SDP.
Resolution: Ensure your PBX is setup to use the WAN IP address for SDP.

Issue: Your PBX and our network cannot agree on a common codec.
Resolution: Ensure that your PBX has G.729 and G.711 codecs enabled.

Issue: Your firewall is blocking outbound RTP packets to IPComms
Resolution: Speaker phones are very prone to echo. Try asking the other party to turn the volume down on their phone. If they are using a speaker phone see if lowering the speaker volume or picking up the handset eliminated the echo.

Issue: You have either not configured an Origination SIP URI for your IPComms SIP Trunk, or have configured a “bad” SIP URI that does not resolve
Resolution: Check that you have configured your SIP URI correctly.

Issue: International call termination is blocked by default by IPComms.
Resolution: Contact IPComms Customer Support, request that your account have international calling enabled, and by completing the Int'l verification process.

Issue: You are attempting to make a call to a country that is not supported by IPComms.
Resolution: Contact IPComms Customer Support and verify available country access.

Issue: The SIP ID and password aren’t correct.
Resolution: Check the SIP ID and password on the phone.

Issue: There are NAT issues on the firewall or in the Asterisk configuration file.
Resolution: Your personal firewall on your network might have NAT (network address translation) enabled, or it might be blocking the ports that need to be open for VoIP to communicate to the outside world. There are two types of traffic that need to be forwarded: SIP signaling and RTP media. The default port for UDP based SIP signaling is port 5060. The RTP media traffic (the actual audio stream) uses a range of 10000-20000.

Issue: The SIP line isn’t registered.
Resolution: Verify your SIP account registration information in your provisioning letter sent in by IPComms upon sign-up.  Ensure it is the same in your device's SIP account settings.

Issue: Your firewall (at your location) is denying SIP requests to IPComms.
Resolution: Open SIP ports 5060 & 5061 on your firewall.

Issue: You are exceeding your SIP trunk concurrent call capacity.
Resolution: Add more ports to your SIP trunk.

Issue:  I just signed up but I haven't received my user name and password.
Resolution: Check your spam folder in your email.  Ensure emails from ipcomms.net are allowed in your spam filter.  Your account will not be activated until we have successfully completed voice verification on the number you provided during sign-up.  Contact customer support to request another voice verification call.