Try using a IVR sound file with 64kbps instead of 128kbps and check.<br><br>Regds<br>Sam<br><br><div class="gmail_quote">On Mon, Dec 20, 2010 at 12:20 AM, Rafqat . <span dir="ltr"><<a href="mailto:rafonline@hotmail.com">rafonline@hotmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
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<br>Hi Sam<br><br>Sorry for the late reply.<br><br>My issue is with only hearing the IVR originating from the switch. <br><br>Phone call debug:<br><br>2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20]<br>
2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20]<br>2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20]<br>
2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]<br>2010-12-19 18:47:16.057209 [DEBUG] sofia_glue.c:2354 Set Codec sofia/internal/<a href="mailto:1006@192.168.0.10" target="_blank">1006@192.168.0.10</a> PCMU/8000 20 ms 160 samples<br>
.<br>.<br>.<br>.<br>EXECUTE sofia/internal/<a href="mailto:1006@192.168.0.10" target="_blank">1006@192.168.0.10</a> sleep(1000)<br>2010-12-19 18:28:22.618376 [DEBUG] sofia.c:4153 Channel sofia/internal/<a href="mailto:1006@192.168.0.10" target="_blank">1006@192.168.0.10</a> entering state [ready][200]<br>
2010-12-19 18:28:22.658376 [DEBUG] switch_rtp.c:2066 Correct ip/port confirmed.<br>EXECUTE sofia/internal/<a href="mailto:1006@192.168.0.10" target="_blank">1006@192.168.0.10</a> voicemail(check default 192.168.0.10)<br>2010-12-19 18:28:23.618435 [DEBUG] switch_ivr_play_say.c:63 No language specified - Using [en]<br>
2010-12-19 18:28:23.818446 [DEBUG] switch_ivr_play_say.c:63 No language specified - Using [en]<br>2010-12-19 18:28:23.818446 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-enter_id.wav] (en:en)<br>2010-12-19 18:28:23.818446 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms<br>
2010-12-19 18:28:25.778567 [DEBUG] switch_ivr_play_say.c:1444 done playing file<br>2010-12-19 18:28:25.878572 [DEBUG] switch_ivr_play_say.c:244 Handle say:[#] (en:en)<br>2010-12-19 18:28:25.878572 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms<br>
2010-12-19 18:28:26.298600 [DEBUG] switch_ivr_play_say.c:1444 done playing file<br><br><br>The phone when playing 'vm-enter_id.wav' seems to jump in volume (initially low volume then high). It also makes a high pitch creaking type noise after 'say:[#] '.<br>
<br>Also when it asks me to leave a message after the tone, it immediatley says 'my message was too short' without allowing me to record anything. Could the phone be sending an echo which causes freeeswitch to think i have ended my recording?<br>
<br>I was wondering if i should invest in some new phones? Any recommendations?<br><br>Cheers<br><br>Raf<br><br><br><br><br><br><br><hr>Date: Sun, 19 Dec 2010 13:10:41 +0530<br>From: <a href="mailto:u2nsam@gmail.com" target="_blank">u2nsam@gmail.com</a><div class="im">
<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br></div>Subject: Re: [Freeswitch-users] audio quality issue<div><div></div><div class="h5"><br><br>
Hi ,<br><br>Do you have issues only with hearing the IVR originating from the switch ? <br>Phone to phone works fine when RTP pass through is enabled ? which codec gets negotiated when IVR plays and what are the IVR file properties which is negotiated?<br>
Your voice mail gets recorded properly means the RTP is passing well to the switch ...<br><br>Regads<br>Sam<br><br><br><div>On Sat, Dec 18, 2010 at 2:52 AM, Rafqat . <span dir="ltr"><<a href="mailto:rafonline@hotmail.com" target="_blank">rafonline@hotmail.com</a>></span> wrote:<br>
<blockquote style="padding-left: 1ex;">
<div>
<br>btw. I have no audio issues when making calls between phones through freeswitch (voice quality is very good).<br><br><br><br><hr>From: <a href="mailto:rafonline@hotmail.com" target="_blank">rafonline@hotmail.com</a><br>
To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>Date: Fri, 17 Dec 2010 21:19:11 +0000<br>Subject: [Freeswitch-users] audio quality issue<div><div></div>
<div><br><br>
<br><br>Hi, <br><br>I recently installed freeswitch 1.0.6 on centos 5.5 (hosted on xen server). I feel as if the audio quality is not as good as what it should be (6-7 out of 10). This is apparent in calling the default voicemail IVR, sometimes I have issues leaving a message, sometimes I hear some weird noises. However, I have no issues with the quality of the recorded voicemail message I leave, it sounds fine on playback.<br>
<br>Is this an issue with my phones (Flexor 500)? I have a softphone which doesn't seem as bad but still has issues.<br><br>I also get the following message on startup of freeswitch:<br><br>2010-12-17 21:17:39.138958 [WARNING] switch_time.c:206 Timer resolution of 4000 microseconds detected!<br>
Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems.<br><br>I assume its related. <br><br>Any help will be much appreciated.<br><br>Cheers<br><br>Raf<br>
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