Fixed Mobile Convergence
From PBXes
Revision as of 20:50, 1 October 2008
Today there are a number of mobile phones available to support VoIP over WLANs and 3G networks. This allows all features of PBXes to be used with one phone from at home, at work and mobile. Advantages over classical telephony are:
- The called party will see the number of the PBX as CallerID, not the mobile phone's number.
- Music on hold instead of annoucements
- Optional time-based routing, call groups, privacy manager
- Calls are billed as data on WLAN/3G plans, not as minutes.
- Avoid high roaming costs by using hotspots when staying in hotels or at airports.
Configuration of such extensions is as follows particularly with regard to seamless switch-over from WLAN/3G to mobile networks:
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Phones
We recommend handsets of the Nokia E or N series where the VoIP client is fully integrated into the phones' firmware (dialing, contacts, etc.). The first models were already introduced in 2006. So these handsets are running very stable today.
If you are going to use WLAN and 3G alternately for VoIP, you should install the 3rd party tool "Psiloc Connect". Choose "always on" and the virtual Psiloc access point for the SIP profile. The same access point can be used for email and position tracking. The tools also has a smart roaming setting.
Psiloc Connect is not applicable if you are going to use VoIP only within WLANs coverage. Then just enter your WLAN access point and the option "always on" for your SIP profile. For email you may enter a list of access points. Position tracking can always connect over 3G because of its low bandwidth usage.
After installation of the tracking client from pbxes.org/m the position is displayed on the webcall page (e.g. for your secretary).
Without speech compression the bandwidth usage is about 1.2 MB per minute. By using compression it can be lowered to 0.3 MB per minute which may be activated per one of the following options:
- Download and install "VoIP settings" from Nokia and choose iLBC codec
- Buy G.729 in the shop (will then be preferable used)
- In Source View/sip.conf add "disallow=all" und "allow=ilbc" (requires PRO account)
PBXes
Add a SIP extension. After registering to it go on fine tuning:
Mobile Device Options
Mobile Option usually remains set to "no".
For ConnectionCheck you have the following options:
- Option A - ConnectionCheck set to "no": This is the default at the lowest bandwidth usage for SIP registration (once per hour). Because all calls will be forked to your mobile the mailbox of your mobile number needs to be de-activated.
- Option B - ConnectionCheck set to "yes": If the mailbox of your mobile number can't be de-activated completely. This option is only available on paid accounts. SIP registration will be done every minute resulting in a total bandwidth usage of about 40 MB monthly.
On paid accounts you can also enable "handoff" in this section. This feature holds calls while switching between networks. Handoff can be triggered manually by muting a call.
Call Forwarding
Forward calls to your mobile number if unavailable or busy. For option A additionally fork calls to your mobile number. If you drop a call ringing on WLAN/3G it will be forwarded to your mobile number.
Trunks
Numbers can be automatically converted from +xxx international format for IP calls by writing 011++|. into the trunk's rules (substitutes + by 011).
Outbound Routing
Depending on the firmware version of your handset you may experience problems on outgoing calls like choppy ring back tone or delayed call establishment after being answered. This can be avoided by turning on the EarlyConnect option which is available to paid accounts.
