Hi, If I understand correctly your requirement: - Asterisk is registered as asterisk@IP - You want to send invites with sip-uri sip:DID@IP instead of the asterisk registration uri (sip:asterisk@IP) Then, you can set the dial-string to the following: <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}}${regex(${sofia_contact(*/${dialed_user})}|^(.+)/sip:.+@(.+)|%1/sip:${sip_dialed_number}@%2)}"/> So when you bridge with user/xxx, fsw will put the DID in the sip-uri and add a Route header sip asterisk reg uri for the next hop. Hope this helps. Best Regards, -- Vallimamod Abdullah SIP Solutions vma@sip.solutions linkedin.com/in/vallimamod .
On 1 Jul 2018, at 22:50, Joseph Waite <joelists@tm.net.uk> wrote:
Hi Alex
This doesn’t seem to do anything, or maybe it is just setting the To: field which I have already got set.
The problem with the To field is that asterisk doesn’t look at this field without customisation which isn’t great for customer ease of use.
Regards
On 29 Jun 2018, at 02:10, Alexey Sibyakin <alex@freeswitch.com> wrote:
Hi,
I'm not sure that I got your scenario right but you can try to do something like this:
<action application="bridge" data="{sip_req_uri=$1}user/foobar"/> (you may need another var, full list is in Confluence)
Regards,
Alex
On Wed, Jun 27, 2018 at 9:33 PM, Joseph Waite <joelists@tm.net.uk> wrote: Hi Guys
Were having issues with Asterisk boxes registering to a trunk on our FreeSwitch box.
We are sending the DID number called down the trunk in the to: field, however Asterisk doesn’t seem to like this.
Is there anyway to manipulate the INVITE to be INVITE sip:DID@ instead of what is currently INVITE sip:user@
Im hoping there is something like sip_invite_to_uri, however google is failing me.
Regards
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_________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
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