Common SIP Problems

You can’t navigate IVR menus. You have poor audio quality on calls. You have echo on calls. Your registration is constantly dropping. You get timeout errors. Your incoming calls don’t reach you. You only get one phone to ring when call hunting. Ahhhhh!! So many possible problems. The reality is that anything that can possibly go wrong may go wrong when using IP based communications. The reverse is that everything can go right. But for the times when they go wrong we’re here to lend a knowledgeable hand.

As a BYOD (Bring Your Own Device) service we have experienced and solved many problems for users of our services. Over time we have become accustomed enough to realize some issues before they occur. VoIP (Voice over IP) may seem scary and unsafe but once you understand where these problems might occur you can avoid them and prepare yourself for a pleasurable and stable communications experience with your Callcentric account. One fact most people may not be aware of is that most of their communications, even on the land line, occur over IP already. We’re here to help and would rather you tackle these issues quickly rather than get frustrated and leave us wondering why. So read on for some of the more common issues we believe can be solved quickly and easily.

REGISTRATION

This is one of the more common problems for users using any SIP (Session Initiation Protocol) based service. There are a few reasons a user may not be able to register:

Incorrect server settings

callcentric.com is the only domain you need. Do NOT use our IP addresses as you will encounter various problems. You do not need to specify a port. If you are required to simply enter 5060. We use DNS (Domain Name System) SRV and can provide many resolvable IPs under a single domain. There’s no need for localized domains. Read more on DNS SRV here.

Wrong user name and/or password

Your username is your 1777 number. Your password is separate from your WEB login password and can be specified independently in your preferences. Make sure you’re not mixing these passwords up.

DNS and network issues

Make sure you can reach http://www.callcentric.com. If you cannot then check your router, network settings and make sure that you can reach other websites before you assume we’re no longer in service ;-).

Firewall

This is a more complicated issue and can be caused by an incorrectly configured firewall or SIP helper applications, most commonly named SIP ALG (Application Layer Gateway). While these features attempt to assist they can cause major problems. Try disabling them if possible. If you are forwarding ports please make sure you are forwarding the proper range to your local network. Some firewalls have a switch to simply enable SIP/VoIP. Otherwise specify port 5060 as an enabled port for outgoing communications.

ISP level issues

Some ISPs restrict SIP and VoIP communications. This is pretty rare these days; however some countries such as the UAE are known for this. Even locally some ISPs may interfere with VoIP communication. Check to make sure that you support SIP on your internet connection by taking tests such as the one here http://myspeed.visualware.com/indexvoip.php.

CALLING

Once you’re registered you will obviously want to make and/or receive calls. Remember that for brand new Callcentric users you can only make and receive calls within the Callcentric network and to and from other SIP users. You can also test incoming and outgoing calling before purchasing anything.

If you have problems placing calls then check out some of the following common problems:

No paid services

We do not add any paid services to your account by default. If you try to place an outgoing call to the PSTN (Public Switched Telephone Network) you will receive error code 1004, described here. This means that you do not have any paid outgoing services on your account. Since your assigned 1777 number is not a real PSTN number you will not be able to receive calls until you add a number/DID to your account.

One way audio and poor call quality

If you have poor call quality on your incoming or outgoing calls then please make sure your codecs are configured properly. Additionally tests such as those from http://myspeed.visualware.com/indexvoip.php should help determine if there is an ISP problem.

Sometimes these problems can be very complicated and may be related to the location you are calling or receiving calls from. This may generally affect international calls. If you are confident the problem is not with your internet configuration or codecs then open a trouble ticket so that we may further assist you.

DTMF

Dual Tone Multi Frequency, or DTMF, is the method by which digital tones, such as numbers, are delivered during a call. You have used DTMF if you have called into your bank and “Pressed 1 for English”. This feature can fail if you have not configured your UA (User Agent) properly. DTMF can also be configured for incoming calls on your IP PBX. Make sure you are using RFC2833/out of band signaling for DTMF tones. You may also enable in-band signaling.

Misconfigured codecs

For calling to the PSTN please make sure you are using the standard voice codecs. you may disable other voice codecs if you do not need them:

PCMU
PCMA
G729A/B

For video calling make sure that both endpoints support the same video codecs. If the same codecs are not supported your video calls will fail. Note that we DO NOT support video calling to the PSTN. Some of the more common video codes are:

H.263
H.263+
H.264

Also make sure that your camera is turned on and nothing is obstructing it.

Broken software or hidden bugs

If you are using a relatively unknown software or hardware please do not expect us to know exactly why a problem is occurring right away. We can work with you to investigate a problem and will inform you of problems we notice. However there are broken software, hardware and aging devices which may stop working at anytime. In these situations we recommend using a known stable solution such as X-Lite, ZoIPer or other SIP UAs you may have lying around to test your problem.

Billing related

DID Forwarding: If you are using a Call Treatment or DID Forwarding rule on your account to forward incoming calls to a mobile or landline number make sure you have enough funds in your balance

Low balance: You may not have enough funds in your balance to receive or place calls when using per minute services

Out of plan minutes: If you have the North America 500 or the North America 1000 then make sure you haven’t exceeded the purchased minutes

Out of plan calling: When using the World Select Residential please be sure that the country you are calling is covered. Note that only some countries include coverage to mobile destinations. Otherwise you will need to add funds to your balance to call these destinations.

OUTGOING CALLING

To further categorize possible calling problems, your outgoing calls may fail for various reasons. You can call our 17771234567 test number, which goes out over the PSTN, to test outgoing calling for free. If you do experience problems then the more immediate reasons may be:

Misconfigured UA dial plans

If you are using a UA which supports dial plans then please make sure your dial plan is configured properly. Some IP PBX solutions are setup to use an access number such as 9. You have a lot of control over things like this and you will want to double check how you’ve set up your outbound calling routes.

Callcentric account settings

Sometimes your calls can be prevented due to some of the preferences on your Callcentric account. Some of the more common ones are:

Allow Calls: You may be calling to an area you have not allowed in your preferences

Simultaneous Calls: You may only be allowing a single call in your preferences, while trying to place multiple calls. If you believe you are experiencing stuck calls place contact support.

INCOMING CALLING

Similarly your incoming calls may fail for various common reasons, including:

Misconfigured inbound routes

If you are using an IP PBX then check your SIP logs/debug logs to see what is happening to your inbound calls. Specifically for Asterisk and trixbox, elastix, pbx-in-a-flash and other Asterisk derived distributions we have setup guides to assist you with this.

Callcentric account settings

Sometimes your incoming calls can fail due to some of the preferences on your Callcentric account. Some of the more common ones are:

DND (Do Not Disturb): Prevents incoming calls

Call waiting: With Call waiting turned off you will only be able to receive one call at once. Turn this off if you plan to receive multiple incoming calls.

Call forwarding settings

If you are forwarding your numbers to another destination then multiple factors could affect your ability to receive calls. These range from incorrect call forwarding rules to problems with the location you are forwarding to. You can review your Call treatment or DID forwarding settings and also make sure the calls reach your account. For this you may view your Calling Reports in your My Callcentric portal. If you believe everything is setup right then open a ticket and we’ll do some investigating on our end.

OVERLY COMPLEX SCENARIOS

We have received enough “master of the network world” issues where users are trying to create very advanced calling setups which go through subnets, routers, firewalls, networks and incompatible services to know that some people will try to get it to work if it’s thinkable. We have no problems with this and welcome user ingenuity; however please understand that certain experiments should be handled on the user end. We can only provide a certain level of support in these cases and cannot spend countless hours dealing with these types of issues. With that said:

Multiple networks

Try not to complicate your setup by placing your UA behind multiple subnets. Ideally your UA should be one level behind your router, or even preferably right behind your gateway if you have an ATA with a built in router. Multiple network levels can cause many problems if not configured right. Additionally it can be frustrating for both support and the end user when a stalemate is reached as to the cause of a problem. If you have one of these scenarios then be sure to point it out when contacting support

Advanced firewalls

A firewall is a staple of internet and networking security. If you are using an advanced firewall with many setting and rules please be sure to try the simplest configuration when experiencing problems. First you will want to make sure that port 5060, or the SIP port you choose, is enabled for incoming traffic. RTP (Real Time Protocol) ports can be specified as well if they are configurable in the UA. Some firewalls enable certain services to be activated on demand. This is useful for the RTP, which is how audio/video, is delivered to you. You may specify a port range or an on demand port range to attempt to combat one-way audio problems.

Incompatible services

Let’s face it, most people don’t know or care what SIP is. They simply know that it’s VoIP and works like Skype so they can place and receive calls. The more knowledgeable of us know what SIP is and may understand that things don’t always work when mixing multiple services. The problem comes when patches are developed to make things work between protocols. Please be aware that patches don’t always work.

We do not advocate against these types of solutions but also would like to point out that we can only assist so far in debugging these types of situations. Please try your best to test your solution independently and, if possible, with multiple providers to make sure it works. We love technology, and we love the hacker philosophy; however we are here to provide support to multiple users and welcome our more advanced users to take that into consideration.

If you are an advanced user then submitting a PCAP or SIP trace of your issue with a detailed explanation can assist us in assisting you very quickly.

These are the most common issues you may experience with Callcentric. They may seem like a lot; however we work in an open BYOD ecosystem and our users are free to use whatever they want with our services. The downside of this is that we have to deal with many different scenarios that may make your entry level tech have nightmares. The upside is that we have professional, intelligent and highly trained staff who are able to tackle most of these issues right from the beginning. We are human and may make mistakes; however we still do work hard to attempt to resolve any problems Callcentric users may have. We hope this information helps our end users and improves their Callcentric experience. As always we are always here to assist if you have a question so open up a trouble ticket and let us help you!

REFERENCE

UA = User Agent
DTMF = Dual Tone Multi-Frequency
IP = Internet Protocol
VoIP = Voice over Internet Protocol
RTP = Real Time Protocol
DNS = Domain Name System
SIP = Session Initiation Protocol
PSTN = Public Switched Telephone Network
ISP = Internet Service Provider
BYOD = Bring Your Own Device