SIP Voice

Project Lead:

rlaager (0 credits)

 

Bounty:

US $0.00(Sponsor Project)
Support standards-compliant SIP voice. Other requirements relating to SIP to PSTN gateways:
 
Note: You can propose changes using the forum below.
Support standards-compliant SIP voice. Other requirements relating to SIP to PSTN gateways:
- This must support making calls to bare phone numbers in addition to SIP URLs. It should do so by appending a default SIP proxy (e.g. 18005551212 becomes 18005551212@sip.example.com). This can probably be implemented in the prpl's normalize function.
- Thought should be given to dialing rules. For example, the numbers 1235551212, 11235551212, and +11235551212 may need to be called as 5551212@sip.example.com. If your current country code is 44, then +11235551212 would need to be dialed as 0011235551212. In the case of centrex lines, an initial 9 might be necessary, except for local extensions. The best way to handle this might be to implement a hook for a plugin.

Suggestions:
This should probably be implemented using Sofia-SIP. It will probably be easier to replace the existing SIP prpl than to bolt on Sofia-SIP.

Change History:
[ACCEPTED] No subject by rlaager on Sun, Aug 23, 2009 @ 02:08 EDT
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