[Freeswitch-users] Manipulate the sip: string in the INVITE header on an outgoing Bridge

[Nathan Downes]

No I didn’t fully read email thought he wanted to change TO: not INVITE:

It was early morning and coffee hadn’t kicked in ☺

If you want to set the invite, don’t bother registering asterisk, just send calls to its IP with a bridge to sofia/external/$1 at ip.add.re.ss:5060<mailto:sofia/external/$1 at ip.add.re.ss:5060>  can do the same in reverse for calls from Asterisk, just have it as an allowed IP for un auth access in FS.

Asterisk will have to accept calls without auth, but then you will get result you want, not sure how well it will work if other side is behind NAT as FS will not do NDLB stuff with this.

Use this for direct routing to other FS instances for some clients.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Alexey Sibyakin
Sent: Tuesday, 3 July 2018 1:48 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Manipulate the sip: string in the INVITE header on an outgoing Bridge

You probably need proper URI as Nathan suggested and not just $1. Also domain should be something that other side can handle.

Regards,

Alex

On Mon, Jul 2, 2018 at 5:50 AM, Joseph Waite <joelists at tm.net.uk<mailto:joelists at 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 at freeswitch.com<mailto:alex at 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 at tm.net.uk<mailto:joelists at 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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--
Alex Sibyakin | Support Engineer
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_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
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_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
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--
Alex Sibyakin | Support Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com<mailto:alex at freeswitch.com>
Website: https://www.FreeSWITCH.com<https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com<mailto:sales at freeswitch.com> for details.
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