Video Walk-through to Call Control/TeXML
Coming soon! This walk-through will demonstrate setting up your Call Control/TeXML Applications.
Configuration of Call Control/TeXML
The Call Control / TeXML applications section is located on the left hand side of the portal
Click on this button below and it will directly get you to the Call Control / TeXML Applications page.
Step by Step Guide to Call Control/TeXML
There are two tabs on this page: Applications and TeXML applications.
We will walk-through both tabs in the following sections.
1. Applications (Call Control App):
App Name:
Click on " Create your first application" and assign a name to this application to better manage the application.
Send a webhook to the URL:
You will need to input a URL where all the webhook events will be sent. Also, you can setup a fail-over URL. If two consecutive delivery attempts to the primary URL fail, Telnyx will attempt delivery to this URL. NOTE: Must include a scheme such as 'https'.
Use Webhook API version:
Determines which webhook format will be used based on the API version V1 or V2.
Enable "hang-up" on timeout:
When enabled, you will specify the number of seconds Telnyx will wait for commands from your application before hanging up.
Custom webhook retry delay (seconds):
In this field, you will need to specify a delay in seconds for Telnyx to wait before retrying an unsuccessful webhook delivery attempt. If not set, Telnyx will retry immediately.
AnchorSite® Selection:
"Latency" directs Telnyx to route media through the site with the lowest round-trip time to the user's connection. Telnyx calculates this time using ICMP ping messages. This can be disabled by specifying a site to handle all media.
DTMF Type:
There are three types in this field: RFC 2833, Inband and SIP INFO.
a. RFC 2833: Default and preferred setting for most use cases, not audible on the call audio.
b. Inband: Digits are passed along just like the rest of your voice as normal audio tones.
c. SIP INFO: Mainly used for SIP to SIP calls. DTMF type is negotiated between parties on the call.
Inbound Settings:
You can configure your global application settings for inbound calls over here.
SIP Subdomain:
Specifies a subdomain that can be used to receive calls to a Connection, in the same way a phone number is used, from a SIP endpoint. Example: the subdomain "example.sip.telnyx.com" can be called from any SIP endpoint by using the SIP URI "sip:@example.sip.telnyx.com" where the user part can be any alphanumeric value. Please note TLS encrypted calls are not allowed for subdomain calls.
SIP Subdomain receive settings:
In this field, either you setup your receive SIP subdomain connection from anyone or only connections.
Inbound Channel Limit:
You can limit the total number of inbound calls to phone numbers associated with this connection.
Outbound settings:
You can configure your global application settings for outbound calls over here.
a. Outbound Voice Profile: Assign your application to an outbound voice profile.
b. Outbound Channel Limit: You can limit the total number of outbound calls to phone numbers associated with this connection.
2. TeXML Applications:
App Name:
Click on " Create your first application" and assign a name to this application to better manage the application.
Voice Method:
In this field, HTTP request method Telnyx will use to interact with your XML Translator webhooks. Either "GET" or "POST".
Send a TeXML webhook to the URL:
You will need to mention a URL where all the XML translator webhook events will be sent. Also, you can setup a fail-over URL. This URL to which Telnyx will deliver your XML Translator webhooks if we get an error response from your "Voice URL"
Status Callback Method:
You will need to mention the HTTP request method Telnyx should use when requesting the "Status Callback" URL.
Send information about call progress events to the URL:
Specify the URL for Telnyx to send requests to containing information about call progress events.
Enable "hang-up" on timeout:
When enabled, you will specify the numbers of seconds to wait for actual application before hanging up.
AnchorSite® Selection:
"Latency" directs Telnyx to route media through the site with the lowest round-trip time to the user's connection. Telnyx calculates this time using ICMP ping messages. This can be disabled by specifying a site to handle all media.
DTMF Type:
There are three types in this field: RFC 2833, Inband and SIP INFO
.
a. RFC 2833: Default and preferred setting for most use cases, not audible on the call audio
b. Inband: Digits are passed along just like the rest of your voice as normal audio tones.
c. SIP INFO: For SIP to SIP calls only. DTMF type is negotiated between the parties on the call
Inbound Settings:
You can configure your global application settings for inbound calls over here.
SIP subdomain:
Specifies a subdomain that can be used to receive calls to a Connection, in the same way a phone number is used, from a SIP endpoint. Example: the subdomain "example.sip.telnyx.com" can be called from any SIP endpoint by using the SIP URI "sip:@example.sip.telnyx.com" where the user part can be any alphanumeric value. Please note TLS encrypted calls are not allowed for subdomain calls.
SIP subdomain receive settings:
In this field, either you setup your receive SIP subdomain connection from anyone or only connections.
Inbound Channel Limit:
You can limit the total number of inbound calls to phone numbers associated with this connection.
Outbound settings:
You can configure your global application settings for outbound calls over here.
a. Outbound Voice Profile: Assign your application to an outbound voice profile.
b. Outbound Channel Limit: You can limit the total number of outbound calls to phone numbers associated with this connection.
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