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



  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.
>
>
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