The Prosody X cards may be used for both TDM and IP networks.
If you are using Prosody X with a TDM network, refer to STG Configuration - Prosody for details as these are the same for Prosody X.
With the v6.4 Aculab drivers, the configuration tool ACT has been extended to support SIP, H323 and IP media stream settings. Also the DSP configuration has been changed to allow the use of a developer provided configuration file which specifies amongst other things all the DSP firmware files that need to be loaded.
You may notice that, when a system has been freshly started, the Prosody X cards are missing from the list. This is because it can take a while for them to boot themselves and then be detected on the network.
Fig. 3 - Prosody X Card Editing
Each card gets it's own IP address and this will be used to send and receive RTP media streams. SIP messages will be sent and received using the host systems normal network connection. In a larger scale production use system, it will be necessary for the Prosody X cards to have IP addresses accessible by all devices from which RTP streams need to be received, i.e. if calls need to be made to external ITSP services the Prosody X cards cannot be behind a NAT router but must have public IP addresses.
Do not check the Use DHCP checkbox.
More information can be found in the Aculab documentation, installed locally, in
C:\Softdial\Aculab\Docs\Prosody_X_card_network_setup_considerations.pdf
Fig. 4 - Prosody X IP Settings 1
Fig. 5 - Prosody X IP Settings 2
Fig. 6 - Prosody X IP Settings 3
Remember to set the configuration for each card.
Fig. 7 - Prosody X TiNG Settings
The Use TRM file checkbox must be ticked and a suitable .trm file selected. The C:\Softdial\STG folder will contain files for both uLaw and ALaw companding modes. Remember to set the configuration for each module on each card as the ACT allows you to have different configurations for each of these. They must all be the same for correct operation of Softdial Telephony Gateway™ (STG).
In a Prosody X environment, STG is able to make use of the Aculab provided SIP services and the RTP media processing capabilities of the boards. The components of the system are:
It is important to use the drivers supplied with STG as other driver versions may not be binary compatible.
With Prosody X, Aculab provide access to SIP services via a virtual port, analogous to an E1/T1 port but without a 23/30 channel limit. The configuration changes for STG to make use of such a port are therefore minimal (the differences are detected and accounted for internally by the software). See the examples below.
This is a typical port configuration section from the STG config.xml file where the board is of Prosody X type:
<portInfos>
<portInfo name="PSTN #1">
<resource>Card/183794/Port/0</resource>
</portInfo>
<portInfo name="PBX #1">
<resource>Card/183794/Port/1</resource>
</portInfo>
<portInfo name="IP #1">
<resource>Card/183794/Port/SIP</resource>
<maxIpCalls>2</maxIpCalls>
</portInfo>
</portInfos>
As you can see, the port number is replaced with the protocol name SIP and the only other addition is a limit on the number simultaneous calls which should be attempted through the port. The named port IP #1 can now be used in a port group definition just as the E1/T1 ports are.
When a call is dialed through a SIP type port, the address needs to be formatted as a SIP URI, e.g.
sip:someone@some.address.net.
In the case of agent logons, this can easily be accomplished by specifying such an address as the Agent Extension (AE) address. A suitable route definition in the config.xml file can be provided to pick out such calls and send them to the required portgroup.
<route name="Internal IP">
<targetPortGroups>
<portGroup>IP</portGroup>
</targetPortGroups>
<acceptList>
<item>sip:</item>
</acceptList>
<originating_addr>sip:plustalk@sytelco.plus.com</originating_addr>
</route>
This is a very simple route definition, simply picking out calls with an address starting with sip: and then routing them to the IP portgroup.
Because the SIP URI's are long, it can be very useful to employ address translations (the addressTranslations element in the config.xml file), since translations are applied to addresses before route processing.
<addressTranslations>
<translate address="601" translated="sip:1002@10.1.1.3"/>
<translate address="701" translated="sip:davidn@proxyserver"/>
<translate address="901" translated="sip:01296381200@sip.plus.net"/>
</addressTranslations>
This allows for the short numbers 601, 701 & 901 to be used in parameters sent to Softdial CallGem™.
It is also possible to register the STG with a SIP proxy server. This can be done with the sipProxy element in the configuration file.
<sipProxy>
<address>sip.plus.net</address>
<port>5060</port>
<protocol>Udp</protocol>
<realm>sytelco.plus.com</realm>
<username>plustalk</username>
<password>*******</password>
</sipProxy>
The realm, username and password elements are optional. The port and protocol default to 5060 and Udp respectively.
Only one proxy server may be registered with, and all calls through all routes have to pass through that proxy.