<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Arial; font-size: 10pt; color: #000000'>Are you still gettint this Problem? I ran into similiar issues, but it's intermittent ...<br><br>best<br>Ray<br><br><hr id="zwchr"><div style="color: rgb(0, 0, 0); font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;"><b>From: </b>"Chris Cureau" <cmcureau@gmail.com><br><b>To: </b>"FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org><br><b>Sent: </b>Wednesday, August 3, 2011 1:28:22 AM<br><b>Subject: </b>Re: [Freeswitch-users] Help with choppy audio after attended transfer<br><br>Latest git doesn't help unfortunately.<br><br>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.<br>
<br><br><br><div class="gmail_quote">On Mon, Aug 1, 2011 at 8:34 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
can you do make current and try latest git?<br>
<div><div></div><div class="h5"><br>
<br>
On Mon, Aug 1, 2011 at 10:46 AM, Chris Cureau <<a href="mailto:cmcureau@gmail.com" target="_blank">cmcureau@gmail.com</a>> wrote:<br>
> Anthony,<br>
><br>
> Thanks for answering...and sorry for the delay. I've already checked all of<br>
> the ptime settings I can, and all phones plus freeswitch are set to use 20ms<br>
> packetization. I've even set "scrooge" in the codec negotiation, but I keep<br>
> running into this issue. I've updated my post with "sofia global siptrace<br>
> on".<br>
><br>
> I am assuming that the ptime issue happens around line 2462<br>
> (<a href="http://pastebin.freeswitch.org/16935" target="_blank">http://pastebin.freeswitch.org/16935</a>)<br>
><br>
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec Compare<br>
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000]<br>
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using PCMU<br>
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf send<br>
> payload to 101<br>
> 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP<br>
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are<br>
> unchanged for sofia/internal/<a href="http://sip:1003@10.0.1.205:5060" target="_blank">sip:1003@10.0.1.205:5060</a>.<br>
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052<br>
> sofia/internal/<a href="http://sip:1003@10.0.1.205:5060" target="_blank">sip:1003@10.0.1.205:5060</a> Setting audio receive payload in<br>
> Re-INVITE to 0<br>
><br>
> Could this be an issue with the Aastra'a firmware? Or maybe the MOH is<br>
> being processed at 30ms instead of 20ms, and the negotiation is not updated<br>
> somehow?<br>
><br>
> I don't mean to sound ignorant, but I'm really at a loss here...and thanks<br>
> again for any help!<br>
><br>
> Cheers,<br>
> Chris<br>
><br>
> On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale<br>
> <<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>> wrote:<br>
>><br>
>> probably ptime related thing.<br>
>> you should have included the sip trace "sofia global siptrace on"<br>
>><br>
>><br>
>> On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <<a href="mailto:cmcureau@gmail.com" target="_blank">cmcureau@gmail.com</a>> wrote:<br>
>> > I'm having some issues with extremely choppy audio after a call has been<br>
>> > sent to another extension via an automated transfer. The audio is great<br>
>> > when the call is answered. Shortly after, the transfer button is<br>
>> > pressed<br>
>> > and the incoming call hears music on hold. The music on hold is sent to<br>
>> > the<br>
>> > caller sounds fine as does the conversation between extensions. When<br>
>> > the<br>
>> > transfer is completed, the caller hears what sounds like someone<br>
>> > speaking<br>
>> > through a fan (though slower) but incoming audio sounds fine.<br>
>> ><br>
>> > Since it's such a large log, I posted it to the FreeSWITCH pastebin:<br>
>> > <a href="http://pastebin.freeswitch.org/16911" target="_blank">http://pastebin.freeswitch.org/16911</a><br>
>> ><br>
>> > I'm thinking that it has something to do with the transition from MOH to<br>
>> > the<br>
>> > internal extension, but I can't figure out what is happening.<br>
>> ><br>
>> > Any ideas?<br>
>> ><br>
>> > _______________________________________________<br>
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>> ><br>
>> ><br>
>><br>
>><br>
>><br>
>> --<br>
>> Anthony Minessale II<br>
>><br>
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>> pstn:<a href="tel:%2B19193869900" target="_blank">+19193869900</a><br>
>><br>
>> _______________________________________________<br>
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><br>
><br>
> _______________________________________________<br>
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><br>
><br>
<br>
<br>
<br>
--<br>
Anthony Minessale II<br>
<br>
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<br>
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<br>
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pstn:<a href="tel:%2B19193869900" target="_blank">+19193869900</a><br>
<br>
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</div></div></blockquote></div><br>
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