Are you still gettint this Problem? I ran into similiar issues, but it's intermittent ...

best
Ray

From: "Chris Cureau" <cmcureau@gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org>
Sent: Wednesday, August 3, 2011 1:28:22 AM
Subject: Re: [Freeswitch-users] Help with choppy audio after attended transfer

Latest git doesn't help unfortunately.

More research is beginning to confirm its a codec issue. When I limit the phone to codes that use 20ms packetization, I hear no incoming audio on outgoing calls. Incoming calls are okay. When using All for codes (and the 30ms packetization) I get the choppy voice when leaving MOH.

On Mon, Aug 1, 2011 at 8:34 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:

can you do make current and try latest git?

On Mon, Aug 1, 2011 at 10:46 AM, Chris Cureau <cmcureau@gmail.com> wrote:
> Anthony,
>
> Thanks for answering...and sorry for the delay. I've already checked all of
> the ptime settings I can, and all phones plus freeswitch are set to use 20ms
> packetization. I've even set "scrooge" in the codec negotiation, but I keep
> running into this issue. I've updated my post with "sofia global siptrace
> on".
>
> I am assuming that the ptime issue happens around line 2462
> (http://pastebin.freeswitch.org/16935)
>
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000]
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using PCMU
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf send
> payload to 101
> 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are
> unchanged for sofia/internal/sip:1003@10.0.1.205:5060.
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052
> sofia/internal/sip:1003@10.0.1.205:5060 Setting audio receive payload in
> Re-INVITE to 0
>
> Could this be an issue with the Aastra'a firmware? Or maybe the MOH is
> being processed at 30ms instead of 20ms, and the negotiation is not updated
> somehow?
>
> I don't mean to sound ignorant, but I'm really at a loss here...and thanks
> again for any help!
>
> Cheers,
> Chris
>
> On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale
> <anthony.minessale@gmail.com> wrote:
>>
>> probably ptime related thing.
>> you should have included the sip trace "sofia global siptrace on"
>>
>>
>> On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <cmcureau@gmail.com> wrote:
>> > I'm having some issues with extremely choppy audio after a call has been
>> > sent to another extension via an automated transfer. The audio is great
>> > when the call is answered. Shortly after, the transfer button is
>> > pressed
>> > and the incoming call hears music on hold. The music on hold is sent to
>> > the
>> > caller sounds fine as does the conversation between extensions. When
>> > the
>> > transfer is completed, the caller hears what sounds like someone
>> > speaking
>> > through a fan (though slower) but incoming audio sounds fine.
>> >
>> > Since it's such a large log, I posted it to the FreeSWITCH pastebin:
>> > http://pastebin.freeswitch.org/16911
>> >
>> > I'm thinking that it has something to do with the transition from MOH to
>> > the
>> > internal extension, but I can't figure out what is happening.
>> >
>> > Any ideas?
>> >
>> > _______________________________________________
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>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale@hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
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>> googletalk:conf+888@conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
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>
>
> _______________________________________________
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> http://www.cluecon.com 877-7-4ACLUE
>
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>
>

--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900

_______________________________________________
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