[Freeswitch-users] Controlling ptime in INVITE

bjordan at e-teleco.com bjordan at e-teleco.com
Mon Dec 31 01:11:02 UTC 2018


I am not familiar if you can mismatch ptimes on A and B legs (I don’t see why it wouldn’t work though I would assume FS would transcode appropriately) but if you set the param on the bridge to that provider it could solve the issue.
<action application="bridge" data="{absolute_codec_string='PCMU at 20i'}sofia/outbound/sip:${sip_to_user}@${outbound_address}"/>

To answer your question you could set it in on the inbound allowed codec by setting the global_codec_prefs in vars.xml, I think you can do the same syntax with the @20i but if you do that in the inbound codecs and the Cisco phone doesn’t renegotiate with your specified ptime it seems possible the call could be lost since there is no valid codec? Maybe someone else could chime in who has more experience with Cisco phones and their behavior.

You could also set it up on the sofia sip profile if you don’t want to do it globally.
https://freeswitch.org/confluence/display/FREESWITCH/Codecs+and+Media#CodecsandMedia-sofia.conf.xmlfile

Thanks,
Branden Jordan

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Joseph Waite
Sent: Friday, December 28, 2018 2:07 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Controlling ptime in INVITE

 I I haven’t.

Is there a way to set this in the inbound allowed codec??
Joe Waite

On 28 Dec 2018, at 20:24, Social Boh <social at bohboh.info<mailto:social at bohboh.info>> wrote:

Hello,

Are you tried to use  absolute_codec_string in the outbound block of your dialplan?

like:

<action application="set" data="absolute_codec_string=PCMU at 20i"/>

Regards

---

I'm SoCIaL, MayBe
El 28/12/2018 a las 11:05, Joseph Waite escribió:
Evening All.

I am having an issue, currently with Cisco SPA phones, where they are trying to use a prime of 30, however one of our providers does not support this and fails the call.

Customer does not notice as call fails over to another provider as it should, however the failover provider is more expensive so not ideal for us.


We don’t notice this unless we examine cdr logs.

Is there a way, either to tell freeswitch to send a response to the phone negotiating time of 20 or to refuse the call if it comes in with time of 30?

Regards



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