https://wiki.freeswitch.org/wiki/Provider_Configuration:_Callcentric

The last bit is what you need.

<!-- There are two other options such as generous, greedy, but scrooge seems to work with CallCentric -->

<param name="inbound-codec-negotiation" value="scrooge"/>




On Sun, Mar 6, 2016 at 7:57 PM, Karl Schmidt <karl@xtronics.com> wrote:
OK - this bit seems likely:


2016-03-06 14:48:04.240958 [WARNING] switch_core_media.c:2042 Asynchronous PTIME not supported,
changing our end from 30 to 20
2016-03-06 14:48:04.260851 [DEBUG] switch_core_media.c:2414 Changing Codec from PCMU@30ms@8000hz to
PCMU@20ms@8000hz

earlier in the log:
Local SDP:
v=0
o=FreeSWITCH 1457269929 1457269930 IN IP4 192.168.1.200
s=FreeSWITCH
c=IN IP4 192.168.1.200
t=0 0
m=audio 27354 RTP/AVP 0 13
a=rtpmap:0 PCMU/8000
a=ptime:30
a=sendrecv
m=audio 27354 RTP/AVP 0 8 9 3 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=sendrecv


So I'm guessing that the PTIME bit is the cause - I found some earlier threads and tried
un-commenting : rtp-autofix-timing=false in sip_profiles/internal.xml


But no joy.

Shouldn't the Asynchronous PTIME  be negotiated before the ringing is set up?

( Call audio is fine - just this ringtone phase on incoming calls )



On 03/05/2016 03:04 PM, Michael Collins wrote:
>
>
> On Fri, Mar 4, 2016 at 3:56 PM, Karl Schmidt <karl@xtronics.com <mailto:karl@xtronics.com>> wrote:
>
>     When people call in (via callcentric) - they hear a stutter in the ringtone.
>
>     My understanding is that the ringtone is generated by freeswitch -  looks like here:
>
>
>     <action application="set" data="transfer_ringback=${us-ring}"/>
>
>
>     We are running the Debian release - 1.4.26~37-1~jessie+1
>
>     Could be this is a new 'feature'? Never sure.
>
>     Anyone else seeing this?
>
>     I'm thinking the ringtone might be using a different CODEX?
>
> You can be absolutely certain of what's happening by taking a pcap of the SIP and RTP that are
> leaving the box when the call comes in from Callcentric. Verify that you're sending a 183 w/SDP and
> that there's actual early media leaving your system. If there is, analyze the RTP and see if the
> stutter can be heard there. If it's PCMU then Wireshark is great for this kind of thing. (Lots of
> info in Confluence on packet captures
> <https://freeswitch.org/confluence/display/FREESWITCH/Packet+Capture> and analysis
> <https://freeswitch.org/confluence/display/FREESWITCH/Wireshark+How+To> if you need a reference.)
>
> The pcap analysis should at the very least help you narrow down where the stutter is coming in to
> play and where to go next.
>
> -MC



--
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Karl Schmidt                                  EMail Karl@xtronics.com
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As a bright friend explained as to why he doesn't watch any TV;
All we have is time - best not to waste it. kps
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