<div dir="ltr"><div>As it turned out the MOH is only working when I initiate then call from the GXP2140.<br></div>When someone else is calling, from another SIP phone or from the PSTN then the MOH ain't working<br><br><br>2015-04-24 17:12:15.192047 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> [BREAK]<br>2015-04-24 17:12:15.192047 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> [BREAK]<br>2015-04-24 17:12:15.212049 [DEBUG] sofia.c:6614 Channel sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> entering state [received][100]<br>2015-04-24 17:12:15.212049 [DEBUG] sofia.c:6624 Remote SDP:<br>v=0<br>o=18 8000 8001 IN IP4 10.0.200.18<br>s=SIP Call<br>c=IN IP4 0.0.0.0<br>t=0 0<br>m=audio 5004 RTP/AVP 0 8 18 9 2 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:9 G722/8000<br>a=rtpmap:2 G726-32/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=sendonly<br>a=ptime:20<br><br>2015-04-24 17:12:15.212049 [DEBUG] switch_channel.c:1787 (sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a>) Callstate Change ACTIVE -> HELD<br>2015-04-24 17:12:15.212049 [DEBUG] switch_core_session.c:1188 Send signal sofia/internal/<a href="mailto:17@192.168.176.6">17@192.168.176.6</a> [BREAK]<br>2015-04-24 17:12:15.232050 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/<a href="mailto:17@192.168.176.6">17@192.168.176.6</a> [BREAK]<br>2015-04-24 17:12:15.352060 [DEBUG] switch_ivr.c:614 sofia/internal/<a href="mailto:17@192.168.176.6">17@192.168.176.6</a> Command Execute playback(/usr/local/freeswitch/sounds/music/output.mp3)<br>EXECUTE sofia/internal/<a href="mailto:17@192.168.176.6">17@192.168.176.6</a> playback(/usr/local/freeswitch/sounds/music/output.mp3)<br>2015-04-24 17:12:15.352060 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@8000hz 1 channels 20ms<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3670 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3670 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [G726-32:2:8000:20:0:1]/[PCMU:0:8000:20:64000:1]<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [G726-32:2:8000:20:0:1]/[PCMA:8:8000:20:64000:1]<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3531 Set telephone-event payload to 101<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:3852 Set 2833 dtmf send payload to 101<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:5100 Audio params changed for sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> from <a href="http://10.0.200.18:5004">10.0.200.18:5004</a> to <a href="http://0.0.0.0:5004">0.0.0.0:5004</a><br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:5111 AUDIO RTP [sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a>] 192.168.176.6 port 23704 -> 0.0.0.0 port 5004 codec: 0 ms: 20<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_media.c:5140 AUDIO RTP CHANGING DEST TO: [<a href="http://0.0.0.0:5004">0.0.0.0:5004</a>]<br>2015-04-24 17:12:15.452067 [DEBUG] sofia.c:7259 Processing updated SDP<br>2015-04-24 17:12:15.452067 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> [BREAK]<br>2015-04-24 17:12:15.472069 [DEBUG] sofia.c:6614 Channel sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> entering state [completed][200]<br>2015-04-24 17:12:15.472069 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> [BREAK]<br>2015-04-24 17:12:15.472069 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> [BREAK]<br>2015-04-24 17:12:15.472069 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> [BREAK]<br>2015-04-24 17:12:15.492071 [DEBUG] sofia.c:6614 Channel sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> entering state [ready][200]<br>2015-04-24 17:12:15.492071 [DEBUG] sofia.c:6624 Remote SDP:<br>v=0<br>o=18 8000 8000 IN IP4 10.0.200.18<br>s=SIP Call<br>c=IN IP4 10.0.200.18<br>t=0 0<br>m=audio 5004 RTP/AVP 0 8 101<br>a=sendrecv<br>a=rtpmap:0 PCMU/8000<br>a=ptime:20<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br><br>2015-04-24 17:12:15.492071 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]<br>2015-04-24 17:12:15.492071 [DEBUG] switch_core_media.c:3670 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match<br>2015-04-24 17:12:15.492071 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]<br>2015-04-24 17:12:15.492071 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]<br>2015-04-24 17:12:15.492071 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]<br>2015-04-24 17:12:15.492071 [DEBUG] switch_core_media.c:3670 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match<br>2015-04-24 17:12:15.492071 [DEBUG] switch_core_media.c:3531 Set telephone-event payload to 101<br>2015-04-24 17:12:15.492071 [DEBUG] switch_core_media.c:3852 Set 2833 dtmf send payload to 101<br>2015-04-24 17:12:15.492071 [DEBUG] sofia.c:7318 Processing updated SDP<br>2015-04-24 17:12:15.492071 [DEBUG] switch_core_media.c:5100 Audio params changed for sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> from <a href="http://0.0.0.0:5004">0.0.0.0:5004</a> to <a href="http://10.0.200.18:5004">10.0.200.18:5004</a><br>2015-04-24 17:12:15.492071 [DEBUG] switch_core_media.c:5111 AUDIO RTP [sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a>] 192.168.176.6 port 23704 -> 10.0.200.18 port 5004 codec: 0 ms: 20<br>2015-04-24 17:12:15.492071 [DEBUG] switch_core_media.c:5140 AUDIO RTP CHANGING DEST TO: [<a href="http://10.0.200.18:5004">10.0.200.18:5004</a>]<br>2015-04-24 17:12:15.752091 [DEBUG] switch_ivr_play_say.c:1747 done playing file /usr/local/freeswitch/sounds/music/output.mp3<br>2015-04-24 17:12:15.752091 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/<a href="mailto:17@192.168.176.6">17@192.168.176.6</a> [BREAK]<br>2015-04-24 17:12:15.752091 [DEBUG] switch_ivr_bridge.c:395 Send signal sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a> [BREAK]<br>2015-04-24 17:12:15.752091 [DEBUG] switch_channel.c:1961 (sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a>) Callstate Change HELD -> UNHELD<br>2015-04-24 17:12:15.752091 [DEBUG] switch_core_io.c:1026 (sofia/internal/<a href="http://sip:18@10.0.200.18:5060">sip:18@10.0.200.18:5060</a>) Callstate Change UNHELD -> ACTIVE<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Apr 23, 2015 at 9:28 PM, Paul Atreides <span dir="ltr"><<a href="mailto:paul.atreides83@googlemail.com" target="_blank">paul.atreides83@googlemail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div>Hi,<br><br></div>I am using a Grandstream GXP2140. When I am having an active call and another person is<br></div><div>calling me then I can switch between the two calls. The person set to hold should hear<br></div><div>the hold music but it only plays the beginning and then just comes silence. I don't get any error messages.<br><br></div><div>How can I set it up that it keeps playing the music?<br></div><div><br>Log:<br>2015-04-23 21:19:00.154957 [DEBUG] switch_channel.c:1961 (sofia/internal/<a href="http://sip:13@10.0.200.13:5060" target="_blank">sip:13@10.0.200.13:5060</a>) Callstate Change HELD -> UNHELD<br>2015-04-23 21:19:00.154957 [DEBUG] switch_core_io.c:1026 (sofia/internal/<a href="http://sip:13@10.0.200.13:5060" target="_blank">sip:13@10.0.200.13:5060</a>) Callstate Change UNHELD -> ACTIVE<br> <br></div><div>Dialplan:<br><action application="bridge_export" data="hold_music=$${sounds_dir}/music/song.mp3"/><br><br></div><div>Thank you<br></div></div>
</blockquote></div><br></div>