I've registered with SIP-trunk providers for experimentation. How can I figure out which codecs a SIP-trunk (gateway in FS parlance?) provides? regards, Oivvio Polite
I believe you should enable siptrace and review the messages sent by the provider. (This is the same method that I used to determine the proper field/information element send by CallCentric containing my DID so that I could route the call properly in my dialplan.) In fs_cli type sofia global siptrace on -- or -- sofia profile sillytel siptrace on /* assuming the sip profile used to reach your provider is named "sillytel"; this eliminates extraneous SIP messages from your trace */ Now place test calls inbound or outbound or both as needed. When you have seen enough type sofia global siptrace off You then have much homework to do :-) --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence
-----Original Message----- From: Oivvio Polite Sent: Tuesday, 26 January, 2016 15:29 Subject: [Freeswitch-users] Checking which codecs a SIP-Trunk provides.
I've registered with SIP-trunk providers for experimentation.
How can I figure out which codecs a SIP-trunk (gateway in FS parlance?) provides?
regards, Oivvio Polite
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On Thu, Jan 28, 2016 at 11:45:54PM -0500, Bote Man wrote:
I believe you should enable siptrace and review the messages sent by the provider. (This is the same method that I used to determine the proper field/information element send by CallCentric containing my DID so that I could route the call properly in my dialplan.)
In fs_cli type
sofia global siptrace on -- or -- sofia profile sillytel siptrace on /* assuming the sip profile used to reach your provider is named "sillytel"; this eliminates extraneous SIP messages from your trace */
Now place test calls inbound or outbound or both as needed.
When you have seen enough type sofia global siptrace off
Thanks! I'll start experimenting with this.
You then have much homework to do :-)
I'm ready do dig in. Oivvio Polite
participants (2)
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Bote Man
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Oivvio Polite