Troubleshooting connections to SIP providers

This topic describes what information you must acquire from your provider and offers tips on configuring SIPSession Initiation Protocol - A communication protocol used for voice and video calls in Internet telephony or private IP telephone systems. interfaces in Kerio Operator.

The sections below cover these topics:

• SIP servers
• Domains
• User names
• Phone numbers
• SIP headers

For more information refer to Connecting to VoIP service providers.

For more information refer to Mapping external and internal numbers.

SIP SRV records

SRV (service) records are entries in DNS that specify the location of service servers. Some SIP providers have SIP SRV records defined for their domain name. Asterisk uses SIP SRV resolution for outbound calls.

To make outbound calls, you must add all proxy servers from your provider's SRV records to your Kerio Operator SIP interfaceAn external interface used for connecting to SIP providers..

To obtain proxies, you can:

• Ask your provider directly.
• Use a Linux dig command.

Example for nexvortex.com:

Command:

dig _sip_udp.nexvortex.com SRV

Result:

_sip._udp.nexvortex.com 1800  IN   SRV    20 0 5060 px5.nexvortex.com.
_sip._udp.nexvortex.com 1800  IN   SRV    30 0 5060 px7.nexvortex.com.
_sip._udp.nexvortex.com 1800  IN   SRV    10 0 5060 px1.nexvortex.com.

To add the proxy servers to the Kerio Operator SIP interface:

1. In the Kerio Operator administration interface, go to Configuration > Call Routing.
2. Double-click the SIP interface.
3. Go to the SIP Details tab.
4. Type the names of the proxy servers in the Inbound proxy field. px5.nexvortex.com, px7.nexvortex.com, and px1.nexvortex.com in the example above.
5. Click OK.

Domains and usernames

If you have issues related to domain or user names while configuring a SIP interface:

• Verify that the Authentication usernames and the SIP name are correct. If you don't get any Authentication username from your provider, assume that they are the same.
• Verify your provider's domain name. Some providers use different terminology, for example, "server name" or "identifier to be used instead of the host name part of the SIP URI".
• Verify that the provider uses the same server name for all SIP server roles (registration, inbound proxy, outbound proxy). If not, configure the Kerio Operator SIP interface correctly for your provider.
• If your provider has multiple SIP servers, type all of them in the Kerio Operator SIP interface.

Phone numbers

Number formats

Before you configure a SIP interface and incoming and outgoing routes in Kerio Operator:

• Verify the format of phone numbers your provider uses.

• The specific number of digits, for example, 9-digit numbers, 10-digit numbers, and so on. note that some US providers use 11-digit numbers instead of 10-digit numbers. In that case, the first digit is always 1.
• E.164 number format, where numbers start with the + sign followed by a country code (for example, +1 in the US). This format often requires configured number rewriting for outgoing calls. For example, rewrite dialed numbers that start with 9 so that the called numbers start with +1 (94084964500 to +14084964500).
• Custom number format, where, for example, providers use the international format without the + sign. Rewrite all national and international numbers to the custom format of the provider.

• Verify that the number format for inbound calls and outbound calls is the same. If not, configure number rewriting correctly for your provider. For example, your provider sends you 10-digit numbers, but requires 11-digit numbers for outbound calls. Create an outgoing route that rewrites numbers to 11-digit format.

Phone numbers as identifiers

Your provider can use phone numbers instead of SIP usernames. If you have only a single external number, many providers use your external number as a SIP usernameAn username for authenticating provided by a SIP provider. as well.

If you have multiple numbers, your provider:

• Uses one of the external numbers as the SIP username.
• Gives you a random string or the common part of your external numbers.
• Gives you multiple accounts, each account with a single external number. Create a SIP interface for each account. You can then assign all interfaces under the same dial-out prefixCountry codes, area codes, a number, or a set of numbers that are dialed before the telephone numbers..

SIP headers

Call setup

When a device initiates a SIP call, it sends the SIP INVITE request. The beginning of the request looks like this:

INVITE sip:13@10.10.1.13 SIP/2.0
Via: SIP/2.0/UDPUser Datagram Protocol - Ensures packet transmission. 10.10.1.99:5060;branch=z9hG4bK343bf628;rport
From: "Test 15" <sip:15@10.10.1.99>tag=as58f4201b
To: <sip:13@10.10.1.13>...

In the request above, extension 15 calls extension 13.

• The called number is in the INVITE (Request-Line in the Kerio Operator administration interface) and To headers.
• The calling number is in the From header.

After you create a SIP interface, Kerio Operator reads the calling number from the From header and the called number from the INVITE header by default. Verify which headers your provider requires and change the settings in the Kerio Operator SIP interface.

For more information refer to Connecting to VoIP service providers.

Example

• You have multiple numbers in range 5551200-5551299.
• Your username is the common part of your numbers, 55512.

The SIP provider sends 55512 in the INVITE header and the specific number, for example 5551234, in the To header.

For outgoing calls, your provider requires 55512 in the From header and the calling number in the P-Preferred-Identity header.

Transferring calls

When transferring calls, Kerio Operator can notify the receiving party about the original caller, otherwise the callee cannot recognize the origin of the call.

Enable the Diversion header on the SIP Details tab of the SIP interface.

If the Diversion header doesn't work, ask your provider which SIP header to use.