asterisk users - Sep 2019 - ptime

We have a customer with a system rejecting calls from Asterisk.  It's
indicating the ptime is 60, but the system admin is saying they only support 20.

They are running asterisk 16.2.1 and using chan_sip
Is there a way to specify this with chan_sip?

Also, for my own curiosity, is there a way to specify this with PJSIP?  (Trying
to migrate customers to PJSIP, but we are holding until asterisk 16.6.0 for a
PJSIP REFER change that has been merged).

Have a great day!
Dan
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On Tue, Sep 3, 2019, at 10:52 AM, Dan Cropp wrote:
>  
> We have a customer with a system rejecting calls from Asterisk. It’s 
> indicating the ptime is 60, but the system admin is saying they only 
> support 20.
> 
> 
> They are running asterisk 16.2.1 and using chan_sip
> 
> Is there a way to specify this with chan_sip?
The ptime is specified the same way in both chan_sip and chan_pjsip, with the
codec. For example:

allow=ulaw:20

I don't know why it would have been offering 60ms though. What codecs were
allowed?

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

Thank you Joshua

We are specifying allow=ulaw

We ran a capture on the asterisk side and it is not specifying a ptime value. 
In the INVITE, asterisk sends a maxptime=150.  Then, the Avaya switch rejects
the call.

We are asking for additional information from the customer and why they think we
are sending ptime=60.


I just ran my own tests with chan_sip and I don't think ptime in the codec
works (at least not for outbound).  With or without the ptime value (allow=ulaw
or allow=ulaw:20) it is not sending ptime in the INVITE packet.  I also tried
changing the ptime in the codec to 40 (just in case it doesn't send if it
matches the default) but it also didn't send it.
Only sending maxptime:150

I did verify this does work with PJSIP.

Dan

-----Original Message-----
From: asterisk-users <asterisk-users-bounces at lists.digium.com> On
Behalf Of Joshua C. Colp
Sent: Tuesday, September 3, 2019 9:02 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] ptime

On Tue, Sep 3, 2019, at 10:52 AM, Dan Cropp wrote:
>  
> We have a customer with a system rejecting calls from Asterisk. It’s 
> indicating the ptime is 60, but the system admin is saying they only 
> support 20.
> 
> 
> They are running asterisk 16.2.1 and using chan_sip
> 
> Is there a way to specify this with chan_sip?
The ptime is specified the same way in both chan_sip and chan_pjsip, with the
codec. For example:

allow=ulaw:20

I don't know why it would have been offering 60ms though. What codecs were
allowed?

--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
www.digium.com & www.asterisk.org

--
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