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 <[email protected]> On Behalf Of Joshua C. Colp Sent: Tuesday, September 3, 2019 9:02 AM To: [email protected] 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
