Changing SDP „a=fmtp:101 0-16“ to „a=fmtp:101 0-15“ request from trunk provider

FreePBX Providers
1

Hello Everyone,

trunk provider for our company wants us to change the outgoing invite fmtp parameter from a=fmtp:101 0-16 to a=fmtp:101 0-15. The reason is that a major telephony group in our country does not accept calls with our current settings.

I searched all the files in Config Editor, SIP settings and Advanced setting in the Web GUI but cannot find anywhere to set up this parameter.

I have some understanding of FreePBX, creating extensions, editing routes etc. but I’m in a situation now, where I have to manage a system that was configured before me and the senior is no longer with the company.

I did some research on my own but am kind of lost now.

All advice would be appreciated.

Thank you,

Tomas

2

There is no configuration option for this. The value is hardcoded into the underlying code, so changing it would require modifying Asterisk and rebuilding it.

3

Thank you for your reply.
Is what you mentioned doable or is it something not recommended? I have a backup of the VM with freepbx, so I could at least give it a try.
Is there some way to go around this, i have DTMF set to RFC2833, would changing it to Info help me to change it in the SIP header?

4

I mean, it’s doable in the sense Asterisk is open source so you can modify the underlying code. Whether it’s doable for you, I’d err on the side of probably not. You would also need to maintain essentially your own version of Asterisk.

INFO DTMF is different, DTMF would no longer be negotiated in SDP and would be sent using SIP INFO requests.

5

Thank you for your input. I´ll have to figure out some other solution or contact the trunk providor.

6

Kamailio can do this, but this is definitely something that your provider should be handling. Their interop with another ITSP is not your problem, the onus is on your provider to make sure your calls get from them to the other group. Its certainly not a “i need to rewrite my pbx’s code or stand up another application” problem for you.

I would tell the provider that it’s not an easy thing to do on your end, and that it’s their problem that they can’t complete your calls, which is what you pay them for

from a technical perspective though, the textops module of Kamailio can make the packet look how they want it to, but that’s not the real solution for the situation

7

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