SIP custom fields

FreePBX Tips and Tricks
1

Hi all,

I’m trying to use an asterisk PBX to connect devices using a customized SIP version.

This devices use additional field both within INVITE method and 200OK.

I’ve experienced that asterisk act as a back to back entity and so re-generates the SIP messages from zero, removing all the custom fields.

Moreover I’ve used SIPAddHeader and I was able to copy and paste the custom fields within the INVITE Method… but my connections con’t comes up because additional headers are “removed” from 200OK message. So I guess SIPAddHeader is not the right way.

Is there a way to make these message pass through asterisk as they are? Moreover is MESSAGES not managed by asterisk?

Thank You in advance.

2

If you hit a square peg with a hammer long enough it will go in a round hole.

3

Again, it is a B2BUA and does NOT “pass through” anything. Again you need a proxy to do what you want, that’s why it is called a proxy :wink: (oh well)

4

dicko

As I wrote in my first comment I was able to pass through all the INVITE custom headers by sipadder commands on extension.conf.
Unfortunately, it seems that SIPAddheader works only on the INVITE method and I have custom headers also on other messages.

I’ll try also on Digium forum to check if there’s any chance, otherwise I’ll change my test bed.

Thank You

5

Asterisk is a B2BUA so nothing is “passing through”, you would need to disassemble the inbound session and use asterisk’s sipaddheader command on the new leg, but this is a FreePBX forum you probably need the Digium Asterisk forum but you should really be using a proxy to do all that.

6

Thank You,

Indeed I know it’s a B2B. I was asking if it exists a particular configuration allowing custom headers passing through.

Massimo

7

Asterisk is a B2BUA not a proxy. It’s supposed to start a new call leg.

It’s just doing what the RFC dictates for the type of device.

It’s not an egg laying milk pig.

8

But the abrasion can hurt on the way in :slight_smile: