dan at amtelco.com
Tue Sep 3 11:19:05 CDT 2019
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.
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:
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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