<div dir="ltr">Hello all,<br><br>I have a problem of "Music on Hold" on AsteriskNow system, based on Asterisk 1.6.2.19 with FreePBX 2.8.1.4<br><br>On another system, when we press the HOLD button on the phone, the phone sends an INVITE with a=sendonly in the SDP, and we get an OK and the system recognizes the a=sendonly request and starts the music on hold, as you can see from the following log:<br>
<br><--- SIP read from UDP:<a href="http://10.0.0.2:5060" target="_blank">10.0.0.2:5060</a> ---><br>INVITE <a href="mailto:sip%3A21@10.0.0.10" target="_blank">sip:21@10.0.0.10</a> SIP/2.0<br>Via: SIP/2.0/UDP 10.0.0.2:5060;rport;branch=z9hG4bK337477455<br>
From: "1001" <<a href="mailto:sip%3A1001@10.0.0.10" target="_blank">sip:1001@10.0.0.10</a>>;tag=446928907<br>To: <<a href="mailto:sip%3A21@10.0.0.10" target="_blank">sip:21@10.0.0.10</a>>;tag=as479a82ac<br>
Call-ID: <a href="mailto:65159842@10.0.0.2" target="_blank">65159842@10.0.0.2</a><br>
CSeq: 22 INVITE<br>Contact: <<a href="http://sip:1001@10.0.0.2:5060" target="_blank">sip:1001@10.0.0.2:5060</a>><br>Max-Forwards: 70<br>User-Agent: sip phone<br>Subject: Phone call<br>Content-Type: application/sdp<br>
Content-Length: 419<br>
<br>v=0<br>o=1001 0000000001 0000000002 IN IP4 10.0.0.2<br>s=A conversation<br>c=IN IP4 0.0.0.0<br>t=0 0<br>m=audio 9000 RTP/AVP 18 4 0 8 23 22 2 21 3 101<br>a=rtpmap:18 G729/8000<br>a=rtpmap:4 G723/8000<br>a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>a=rtpmap:23 G726-16/8000<br>a=rtpmap:22 G726-24/8000<br>a=rtpmap:2 G726-32/8000<br>a=rtpmap:21 G726-40/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=sendonly<br>
<br><-------------><br>--- (12 headers 18 lines) ---<br>Sending to 10.0.0.2 : 5060 (no NAT)<br>Found RTP audio format 18<br>Found RTP audio format 4<br>Found RTP audio format 0<br>Found RTP audio format 8<br>Found RTP audio format 23<br>
Found RTP audio format 22<br>Found RTP audio format 2<br>Found RTP audio format 21<br>Found RTP audio format 3<br>Found RTP audio format 101<br>Found audio description format G729 for ID 18<br>Found audio description format G723 for ID 4<br>
Found audio description format PCMU for ID 0<br>Found audio description format PCMA for ID 8<br>Found audio description format G726-16 for ID 23<br>Found audio description format G726-24 for ID 22<br>Found audio description format G726-32 for ID 2<br>
Found audio description format G726-40 for ID 21<br>Found audio description format GSM for ID 3<br>Found audio description format telephone-event for ID 101<br>Capabilities: us - 0x28010e (gsm|ulaw|alaw|g729|h263|h264), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>Peer audio RTP is at port <a href="http://0.0.0.0:9000" target="_blank">0.0.0.0:9000</a><br>Peer doesn't provide video<br>
<br><--- Transmitting (no NAT) to <a href="http://10.0.0.2:5060" target="_blank">10.0.0.2:5060</a> ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK337477455;received=10.0.0.2;rport=5060<br>
From: "1001" <<a href="mailto:sip%3A1001@10.0.0.10" target="_blank">sip:1001@10.0.0.10</a>>;tag=446928907<br>
To: <<a href="mailto:sip%3A21@10.0.0.10" target="_blank">sip:21@10.0.0.10</a>>;tag=as479a82ac<br>Call-ID: <a href="mailto:65159842@10.0.0.2" target="_blank">65159842@10.0.0.2</a><br>CSeq: 22 INVITE<br>Server: PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces<br>Contact: <<a href="mailto:sip%3A21@10.0.0.10" target="_blank">sip:21@10.0.0.10</a>><br>Content-Length: 0<br><br><------------><br>Audio is at 10.0.0.10 port 10022<br>Adding codec 0x100 (g729) to SDP<br>
Adding codec 0x4 (ulaw) to SDP<br>Adding codec 0x8 (alaw) to SDP<br>Adding codec 0x2 (gsm) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br><br><--- Reliably Transmitting (no NAT) to <a href="http://10.0.0.2:5060" target="_blank">10.0.0.2:5060</a> ---><br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK337477455;received=10.0.0.2;rport=5060<br>From: "1001" <<a href="mailto:sip%3A1001@10.0.0.10" target="_blank">sip:1001@10.0.0.10</a>>;tag=446928907<br>
To: <<a href="mailto:sip%3A21@10.0.0.10" target="_blank">sip:21@10.0.0.10</a>>;tag=as479a82ac<br>
Call-ID: <a href="mailto:65159842@10.0.0.2" target="_blank">65159842@10.0.0.2</a><br>CSeq: 22 INVITE<br>Server: PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces<br>Contact: <<a href="mailto:sip%3A21@10.0.0.10" target="_blank">sip:21@10.0.0.10</a>><br>
Content-Type: application/sdp<br>Content-Length: 340<br><br>v=0<br>o=root 891217183 891217184 IN IP4 10.0.0.10<br>s=PBX<br>c=IN IP4 10.0.0.10<br>t=0 0<br>m=audio 10022 RTP/AVP 18 0 8 3 101<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>
a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=recvonly<br><br><------------><br> -- Started music on hold, class 'default', on SIP/21-00000da4<br>
<br><br>On the AsteriskNow system, it gives an OK, but nothing happens, there's no music and after some time, the call even drops for empty RTP. That's the log there:<br><br><--- SIP read from UDP:<a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>
INVITE <a href="mailto:sip%3A200@192.168.1.10">sip:200@192.168.1.10</a> SIP/2.0<br>From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8<br>
To: "200"<<a href="mailto:sip%3A200@192.168.1.10">sip:200@192.168.1.10</a>>;tag=as6b718821<br>Call-ID: <a href="mailto:1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10">1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10</a><br>
CSeq: 1 INVITE<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-81eef-1fb8d69c-75bdf605<br>Max-Forwards: 70<br>Supported: replaces,100rel<br>User-Agent: SIP Phone<br>Contact: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>><br>
Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE<br>Content-Type: application/sdp<br>Content-Length: 369<br><br>v=0<br>o=500 2146032705 0 IN IP4 192.168.1.109<br>s=SIPPhone Session<br>i=Audio Session<br>c=IN IP4 0.0.0.0<br>
t=0 0<br>m=audio 16384 RTP/AVP 18 8 0 18 4 9 101<br>a=sendonly<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:18 G729/8000<br>a=rtpmap:4 G723/8000<br>a=rtpmap:9 G722/16000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-11<br><br><-------------><br>--- (13 headers 17 lines) ---<br>Sending to 192.168.1.109 : 5060 (NAT)<br><br><--- Transmitting (NAT) to <a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>
SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-81eef-1fb8d69c-75bdf605;received=192.168.1.109<br>From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8<br>
To: "200"<<a href="mailto:sip%3A200@192.168.1.10">sip:200@192.168.1.10</a>>;tag=as6b718821<br>Call-ID: <a href="mailto:1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10">1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10</a><br>
CSeq: 1 INVITE<br>Server: FPBX-2.8.1(1.6.2.19)<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Contact: <<a href="mailto:sip%3A200@192.168.1.10">sip:200@192.168.1.10</a>><br>
Content-Length: 0<br><br><br><------------><br>Audio is at 192.168.1.10 port 18380<br>Adding codec 0x8 (alaw) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br><br><--- Reliably Transmitting (NAT) to <a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-81eef-1fb8d69c-75bdf605;received=192.168.1.109<br>From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8<br>
To: "200"<<a href="mailto:sip%3A200@192.168.1.10">sip:200@192.168.1.10</a>>;tag=as6b718821<br>Call-ID: <a href="mailto:1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10">1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10</a><br>
CSeq: 1 INVITE<br>Server: FPBX-2.8.1(1.6.2.19)<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Contact: <<a href="mailto:sip%3A200@192.168.1.10">sip:200@192.168.1.10</a>><br>
Content-Type: application/sdp<br>Content-Length: 235<br><br>v=0<br>o=root 656809389 656809389 IN IP4 192.168.1.10<br>s=Asterisk PBX 1.6.2.19<br>c=IN IP4 192.168.1.10<br>t=0 0<br>m=audio 18380 RTP/AVP 8 101<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br>a=sendrecv<br><br><------------><br><br><--- SIP read from UDP:<a href="http://192.168.1.109:5060">192.168.1.109:5060</a> ---><br>ACK <a href="mailto:sip%3A200@192.168.1.10">sip:200@192.168.1.10</a> SIP/2.0<br>
From: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>>;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8<br>To: "200"<<a href="mailto:sip%3A200@192.168.1.10">sip:200@192.168.1.10</a>>;tag=as6b718821<br>
Call-ID: <a href="mailto:1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10">1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10</a><br>CSeq: 1 ACK<br>Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-81eef-1fb8d6e2-346b6e2d<br>Max-Forwards: 70<br>
User-Agent: SIP Phone<br>Contact: <<a href="http://sip:500@192.168.1.109:5060">sip:500@192.168.1.109:5060</a>><br>Content-Length: 0<br><br><br>The SIP peer is set to canreinvite (if it matters).<br><br>Does anyone know why it doesn't start the MOH process on this system, unlike the other one?<br>
<br>Thanks,<br><br>Michael<br></div>