> Hi > > Click-to-call sends an initial INVITE with an SDP offer that > only includes G.711u as audio codec. Once this call is > answered, a REFER is then sent which effectively blind > transfers the call to the remote party. During the transfer > the two parties can negotiate new media however if the > initial INVITE fails, of course the REFER never actually happens. > > The endpoints we use support G.711u it is just a policy to > use G.711a (given it is the national standard here in New > Zealand, like Europe). I can enable G.711u at a lower > priority in the endpoints - that's easy enough - I was just > curious as to whether or not it can be controlled by a > configuration parameter on SIPXecs. It doesn't use the Media > server codec settings, which I guess makes sense since it is > not actually ever sending any media.
This cannot be configured anywhere. Currently, your only option is to enable G.711u on your system. Going forward, the best way to solve this problem would be for the 3PCC to offer g.711u, g.711a and g.729 in its SDP to increase the odds that the offer will be acceptable to the endpoint. Are you able to raise a tracker for this? > > Regards > Justin > > > On Fri, Jun 11, 2010 at 9:30 PM, Tony Graziano > <[email protected]> wrote: > > > As I understand this feature, click to call uses > "REFER", and is not > designed like a 3PCC B2BUA. Once the invite goes out the > phones/endpoints involve negotiate the media based on > an offering. > Since click-to-call sends the refer, the "INVITE" to > your phone does > not have media involved until it is negotiated with the > called party. > > Can you be more specific as to the problems you are > having and/or what > type of endpoint you use? Since the Media Server, > trunks and the like > all also use G711uLaw, I think it would be best if you > could better > describe your issue. Do calls actually fail? > > > On Fri, Jun 11, 2010 at 3:30 AM, Justin Menga > <[email protected]> wrote: > > Hi > > I have a small issue with Click-to-Call from > Phonebook in SIPXecs. Problem > > is that SIPXecs always sends a proposal for only > G.711u in SIP INVITE to the > > "calling" party. My deployment uses G.711a and > ideally I'd like to be able > > to control this if possible. > > Any way to change this? > > Thanks > > Justin > > > _______________________________________________ > > sipx-users mailing list [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipx-users > > Unsubscribe: > http://list.sipfoundry.org/mailman/listinfo/sipx-users > > sipXecs IP PBX -- http://www.sipfoundry.org/ > > > > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > > > > _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
