[OpenSIPS-Users] SIP Pointers

[James Mbuthia]

Hi,

Am a SIP newbie and I need some help, this isn't an opensips question, I
just need some guidance and pointers and I hope someone can help me.

I am developing a web-basedsip softphone using Opensips as the proxy.
Eventually I want to integrate the proxy with Asterisk for PSTN calling. My
client is working fine and a 3 way handshake is successfully done. What do I
need to do to carry the call from the UAC to the UAS once the session has
been setup? Will I need to license codecs to handle the call? As I
mentioned, this isn't about Opensips, just need some pointers and guidance.
Any help will be appreaciated.

regards,
james
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.opensips.org/pipermail/users/attachments/20101026/36afe156/attachment.htm