I have completed a few more changes to the package.
1. When adding a Gateway the following command is now called automatically through a socket connection.
sofia profile external rescan reloadxml
This will pick up new gateways that have been added or deleted and handle them appropriately without restarting FreeSWITCH. There still may be times a few times that require a restart for a gateway. I could have been more aggressive and had it run: sofia profile external restart reloadxml but that would interrupt current incoming/outgoing calls so I chose not to go that way.
2. Added a few more input fields for Gateways such as 'Realm' and 2 others fields in many cases this is not needed but I wanted to make sure that there is flexibility in GUI for special situations. If a field is left blank then the corresponding xml parameter tag will not be used in the config. The fields under the 'Gateways' tab match up exactly with the SIP provider examples at:
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples3. The 'Modules' tab has been changed from the original text area to a list in the GUI that allows you to enable or disable different modules. Note: not all modules are available in this build... an example would be Cepstral text to speech which doesn't have a native FreeBSD build. Flite text to speech is available if its modules is enabled.
Now to answer your questions...
Sounds good. I confirmed that the reloadxml command is working, although it didn't work for me when adding extensions immediately after install, but that might just be FreeSwitch trying to be efficient and not having Sofia loaded (since I didn't have any sip devices listed prior to that), or who knows. It's working now, so whatever.
Sofia (SIP) is loaded by default. If you restart FreeSWITCH under Services then you should check the Status tab to make sure that the sofia internal profile is loaded if it does not load it will say 'Invalid Profile!' This is caused from a port not closing by the time you restart FreeSWITCH. It can be solved by shutting down the service longer sometimes up to 2-5 minutes then starting it again. Rebooting will also work to clear that up.
Another question...you'll be getting several package bug reports from me (if there are any bugs and I'm not just being retarded.
), do you prefer I PM them to you?
If the information can be valuable to others then the forum. Items that are not likely to be useful to others you can pm me.
#1: When changing the area code in the settings tab, 7 digit calls still get prepended with 918, due to the <X-PRE-PROCESS cmd="set" data="default_areacode=918"/> line in the VAR tab. Should this be changed to the value from the Settings tab? EDIT: I actually put $${default_areacode} in the area code field on the settings page...I figure that should pull it from the vars.xml file, where I have it set correctly. Is this correct? Also, what xml setting does the "area code" on the settings page change?
$${default_areacode} in the area code field on the settings page should work fine.
The settings page populates the area code in the following xml file /usr/local/freeswitch/conf/directory/default/default.xml
#2: In /usr/local/freeswitch/conf/dialplan/default/01_provider.com.xml, 10 digit calls just drop. Adding the following section will fix that by prepending a 1, and sending the call out.
<extension name="domestic.provider.com">
<condition field="${toll_allow}" expression="domestic"/>
<condition field="destination_number" expression="^(\d{10})$">
<action application="set" data="effective_caller_id_number=$${outbound_caller_id_number}"/>
<action application="set" data="effective_caller_id_name=$${outbound_caller_id_name}"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/1$1"/>
</condition>
</extension>
This can also be done from the dialplan tab as noted in the this first message in this thread. As you have shown the caller id can also be done for all calls going outbound to the gateway with the 10 digits shown above. The other method for caller id is per extension which can be done by setting the effective caller id on the 'Extension' tab.
#3: I can't get the "say:" command in the IVR xml file to work. I get the following error in the console:
2008-10-28 22:43:15 [ERR] mod_native_file.c:68 native_file_file_open() Error opening /usr/local/freeswitch/sounds/en/us/callie/say:Press 1 to join the conference, Press 2 to join the other conference.PCMU
I have not worked with the XML IVR because I prefer doing the IVR in javascript. For help with the XML IVR see the wiki at
http://wiki.freeswitch.org/To see an example IVR in javascript.
http://wiki.freeswitch.org/wiki/Javascript_Examples look for the example ivrmenuofficehours.js link.
The sounds under /usr/local/freeswitch/sounds/en/us/callie/ are pre-recorded messages using Cepstral's text to speech engine.
At this time Cepstral does not have a native FreeBSD build. Alternatives are to pre-record messages on a different computer, use Flight text to speech engine, or make your own voice recordings.
#4: In /usr/local/freeswitch/conf/dialplan/default/01_provider.com.xml all the caller id setup is done with single dollar signs...IE, ${outbound_caller_id_number}. I'm no FreeSwitch guru, so I can't explain why, but I can't pass callerid through Voicepulse to the PSTN until I change them all to double dollar signs, as I saw in the vars.xml file...note that I didn't add a dollar sign to ${default_gateway}, only the caller_id settings, and that's apparently working...I'm not sure what the difference is.
Looks like you found a missing $ in the default config. Next time I compile I will check to see if that has been corrected if not I will report it if you don't beat me to it.
#5: In /usr/local/freeswitch/conf/dialplan/default/01_provider.com.xml all the caller id variables reference $${outbound_caller_id_number} or =$${outbound_caller_id_name}. In the vars.xml file, the only caller id variables are outbound_caller_id and outbound_caller_name. Until I assign my caller id data to the first variables, outbound_caller_id_number and outbound_caller_id_name, I can't pass callerid through Voicepulse to the PSTN.
Vars.xml maps to the 'Vars' tab in the GUI. It sets up variables that can be used anywhere in the config. There may be some cases where the vairables are not being used anywhere in the config. FreeSWITCH is a young open source project and the configs are evolving and being improved over time.
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