I am also fighting the same problem as Gabriel -i.e., I can't get a call from outside to forward back to the outside on Speakeasy's SIP trunks.
I wrote to Speakeasy asking them if they support P-Asserted-Identity. Here is the answer I got: **** We do not support P-Asserted Identity at this time. We are currently using Remote Party ID for privacy. The FROM field of the SIP INVITE must be a registered phone number or provide Remote Party ID. This also applies for any call forwarding, blind xfrs, etc. Many IP-PBXs may not support Remote Party ID, in which case it's suggested you use a Caller ID override. **** Any ideas on what we could do to get our forwards to work? Thanks On Jan 25, 2010, at 7:54 AM, M. Ranganathan wrote: > On Mon, Jan 25, 2010 at 4:34 AM, gabriel <g...@bayintegrated.net> wrote: >> >> I have the following problem with the call fwd. >> >> If I set up a user so that the calls to his extension gets fwded to his >> cell, that feature works only when dialing from other extensions. In that >> case it works as designed and the cid received is the correct one (the cid >> assigned to the internal user that originally called) >> >> After reading the very good doc it appears that it has to do with >> P-Asserted-Identity, (sipx tries to fwd the incoming caller ID out to the >> sip trunk and the ITSP doesn't accept/doesn't know how to handle it) >> >> so the q is if I uncheck "Use default asserted identity " then I am not >> sure what am I supposed to use in the "Asserted identity" field (having >> speakeasy as ITSP) >> >> is there another way to fix this ? I should say that I have a default >> caller ID enabled for the SIP trunk to be used in the case that an >> internal user calling out doesn't have one assigned >> >> is it really that they don't support this or am I doing something wrong ? >> >> -gabriel > > > You can try leaving P-Asserted-Identity blank and deselect "use > default asserted identity". In that case the P-A-I header will not be > used. > > Ranga >> >> >> >> On Sun, 24 Jan 2010, Pizza Napoletana wrote: >> >>> On Jan 24, 2010, at 6:52 PM, M. Ranganathan wrote: >>>>> But Speakeasy gave me a whole bunch of parameters when they provisioned >>>>> the trunks. Here is what they gave (which I think is for asterisk): >>>>> ... >>>>> insecure=very >>>> No idea what "insecure=very" means but that does sound frightening. :-) >>> >>> Per * doc, "insecure=very" means "To allow registered hosts to call without >>> re-authenticating". >>> >>>>> qualify=no >>> >>> Per * doc, qualify=yes means * will send a SIP OPTIONS command every few >>> seconds to check that the device is still online. >>> >>>>> type=peer >>> >>> Per * doc, "type=peer" means a SIP entity to which Asterisk sends calls >>> (versus a "user" who receives calls, or a "friend" that does both). >>> Confusing to me! >>> >>>> Try a few call flows. I suspect these are not relevant. If in and >>>> outbound calling are working then chances are that you are OK. >>> >>> Great. Thanks. I'll do some rigorous testing tomorrow. >>> >>> >>> _______________________________________________ >>> sipx-users mailing list sipx-users@list.sipfoundry.org >>> 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/ >>> >> _______________________________________________ >> sipx-users mailing list sipx-users@list.sipfoundry.org >> 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/ >> > > > > -- > M. Ranganathan _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org 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/
