From VoIP.ms Wiki
You may need to add from domain param set to voip.ms for termination to work.
<param name="from-domain" value="voip.ms"/>
- If you have a linksys device (spa2102 spa5xx series), they will reject calls if ptime is not set to 20. Make sure you change that in the phone's configuration (rtp packet size 0.020 [from 0.030]).
- Based on icall config above
- Only tested with IP auth
- Place the following in conf/sip_profiles/external/voipms.xml (Please note for some versions "sip_profiles" has been changed to "vanilla/sip_profiles"):
<include> <gateway name="voipms"> <!-- Replace the values below with your Voip.ms username and password. --> <param name="username" value="your_username" /> <param name="password" value="your_password" /> <!-- This gateway could be different depending on which switch you are on --> <param name="proxy" value="montreal|houston|newyork|etc.voip.ms" /> <param name="realm" value="voip.ms" /> <!-- This should be set to "true" for registration based --> <param name="register" value="true" /> <!-- Voip.ms requires the Remote-Party-Identity Header to be set in the Sip invite for Caller-ID to work right DON'T FORGET TO REMOVE ANY CALLER ID INFO IN http://voip.ms->Main Menu->Account Settings->General->CallerID Number --> <param name="sip_cid_type" value="rpid" /> <!--Setting in one place is much easier than everywhere you may bridge. You can do this since 2010 Sept 27 http://jira.freeswitch.org/browse/FS-2722 --> </gateway> </include>
Voip.ms requires the Remote-Party-Identity Header to be set in the Sip invite. Use:
<param name="sip_cid_type" value="rpid" />
- destination_number (inbound public) is getting set as my Sip_profile username.
- if you have a subaccount set up, make sure "Device Type" is set to "Asterisk, ip pbx, gateway or voip switch" and not "ata device, ip phone or soft phone". Once this is fixed, destination_number will be the correct DID number.
- Configure your firewall to allow incoming SIP and RTP traffic from the IP address of your voip.ms “DID POP”. SIP is TCP/UDP port 5060. For RTP voip.ms uses UDP ports 10001:20000. Make sure that the RTP start and end ports in autoload_configs/switch.conf.xml match these values.
- Configure NAT. You need to forward the SIP and RTP traffic via NAT to your FreeSwitch server IP. For SIP traffic you will also need to change the destination port from TCP/UDP 5060 to TCP/UDP 5080. FreeSwitch listens for external connections on port 5080.
- Restart FreeSwitch. Run a recursive chown to make sure that the freeswitch user owns these new files. I have found FreeSwitch to be tricky when it comes to reloading configurations. Usually a reloadxml in the CLI will work but sometimes you also have to do a sofia rescan. I recommend you restart FreeSwitch after making these changes.
- Test Configuration. Once FreeSwitch has restarted (this takes a few minutes) launch fs_cli and check the SIP connections status with sofia status. If all went well you should see something like this: external::voipms gateway sip:[acct #]@[pop].voip.ms REGED