Modifying To parameters on the SIP Invite

Asterisk Asterisk Support
1

Hey there,

I’m running Asterisk 1.6 with FreePBX.

I need to be able to set the npdi and rn parameters in the To header on my SIP invites, however I can’t seem to get it working.

Using this code below in extentions_custom.conf:

[macro-dialout-trunk-predial-hook]
exten => s,1,Set(__SIP_URI_OPTIONS=npdi=yes\;rn=8082900191)
exten => s,n,MacroExit()

I can set the To header as such:

To: <sip:8081010007@172.22.135.22;npdi=yes;rn=8082900191>;tag=blah

However I believe what I need to do is this:

To: <sip:8081010007@172.22.135.22>;npdi=yes;rn=8082900191;tag=blah

Is this even possible? And if so can anybody help me do it?

Thanks

2

SIP_URI_OPTIONS won’t work the way you’d like. You’ll have to dive into the code to change it.

3

Hey Malcolm,

Thanks for your speedy reply - do you mean I would need to dive into the asterisk code itself (as opposed to modifying this in the *.conf files).

Thanks,
Laurence

4

Yes. You cannot change the functionality of that SIP channel variable by editing configuration files. You’ll have to edit the source.

5

Hey Malcolm,

Thanks for your help - turns out my understanding of the rn and npdi parameters was incorrect and what I wanted to send was:

INVITE sip:8081010001;npdi=yes;rn=8081010000@172.22.135.22 SIP/2.0

In case anybody wants to do the same -

I still set this in extentions_custom.conf:

[macro-dialout-trunk-predial-hook]
exten => s,1,Set(__SIP_URI_OPTIONS=npdi=yes\;rn=8082900191)
exten => s,n,MacroExit()

I then made some minor changes to chan_sip.c so that SIP_URI_OPTIONS was added before the @ in the request URI.

Thanks for your help.
Laurence

6

Would you share what changes were made to chan_sip.c?