[asterisk-dev] chan_sip usereqphone option

[Pavel Troller]

Hi!

> 
> On 11 Jun 2014, at 19:35, George Joseph <george.joseph at fairview5.com> wrote:
> 
> > On Wed, Jun 11, 2014 at 12:24 PM, Mark Michelson <mmichelson at digium.com> wrote:
> > Hi!
> > 
> > It has been brought to my attention that chan_pjsip does not have an equivalent to chan_sip's usereqphone option. However, nobody here seems to know how useful such an option actually would be. The result is, we have no idea if an equivalent should be made in chan_pjsip or if we should just leave that option alone. I figure that the -dev list may have some experience with setting up accounts with a variety of SIP providers and could shed some light on the usefulness of that option.
> > 
> > Thanks.
> > 
> > I've set up a dozen providers but never needed usereqphone.   For pjsip though couldn't you just add ';user=phone' to a contact uri if you needed it?
> > 
> I've seen quite a few SIP trunks that require this. But not in a contact.
> 
> /O
> 
This information is there to specify, that the user field of the URI (in any of
headers specifying an address) contains an E.164 formatted telephone number.
When not present, sip servers may omit E.164 parsing and conversion rules (for
example, substitution of "+" to "00" or other prefix, deleting the country 
code for local calls etc.), thus making the calls unroutable or containing
irregular information (i.e. calling party number in incorrect format). However,
it CANNOT be present for URIs, which don't contain the number (like
anonymous at anonymous.invalid). Proper implementation of this tag is necessary
for interworking of Asterisk with practicaly any modern professional NGN
equipment (Application Servers, IMS network elements etc.).
  So, please, full implementation (like in chan_sip) is really necessary.

  WIth regards,
    Pavel