Hello all, I have a problem of "Music on Hold" on AsteriskNow system, based on Asterisk 1.6.2.19 with FreePBX 2.8.1.4
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: <--- SIP read from UDP:10.0.0.2:5060 ---> INVITE sip:21@10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:5060;rport;branch=z9hG4bK337477455 From: "1001" <sip:1001@10.0.0.10>;tag=446928907 To: <sip:21@10.0.0.10>;tag=as479a82ac Call-ID: 65159842@10.0.0.2 CSeq: 22 INVITE Contact: <sip:1001@10.0.0.2:5060> Max-Forwards: 70 User-Agent: sip phone Subject: Phone call Content-Type: application/sdp Content-Length: 419 v=0 o=1001 0000000001 0000000002 IN IP4 10.0.0.2 s=A conversation c=IN IP4 0.0.0.0 t=0 0 m=audio 9000 RTP/AVP 18 4 0 8 23 22 2 21 3 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:23 G726-16/8000 a=rtpmap:22 G726-24/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:21 G726-40/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly <-------------> --- (12 headers 18 lines) --- Sending to 10.0.0.2 : 5060 (no NAT) Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 23 Found RTP audio format 22 Found RTP audio format 2 Found RTP audio format 21 Found RTP audio format 3 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G726-16 for ID 23 Found audio description format G726-24 for ID 22 Found audio description format G726-32 for ID 2 Found audio description format G726-40 for ID 21 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 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) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 0.0.0.0:9000 Peer doesn't provide video <--- Transmitting (no NAT) to 10.0.0.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.2:5060 ;branch=z9hG4bK337477455;received=10.0.0.2;rport=5060 From: "1001" <sip:1001@10.0.0.10>;tag=446928907 To: <sip:21@10.0.0.10>;tag=as479a82ac Call-ID: 65159842@10.0.0.2 CSeq: 22 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:21@10.0.0.10> Content-Length: 0 <------------> Audio is at 10.0.0.10 port 10022 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.0.0.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.2:5060 ;branch=z9hG4bK337477455;received=10.0.0.2;rport=5060 From: "1001" <sip:1001@10.0.0.10>;tag=446928907 To: <sip:21@10.0.0.10>;tag=as479a82ac Call-ID: 65159842@10.0.0.2 CSeq: 22 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:21@10.0.0.10> Content-Type: application/sdp Content-Length: 340 v=0 o=root 891217183 891217184 IN IP4 10.0.0.10 s=PBX c=IN IP4 10.0.0.10 t=0 0 m=audio 10022 RTP/AVP 18 0 8 3 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on SIP/21-00000da4 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: <--- SIP read from UDP:192.168.1.109:5060 ---> INVITE sip:200@192.168.1.10 SIP/2.0 From: <sip:500@192.168.1.109:5060 >;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8 To: "200"<sip:200@192.168.1.10>;tag=as6b718821 Call-ID: 1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-81eef-1fb8d69c-75bdf605 Max-Forwards: 70 Supported: replaces,100rel User-Agent: SIP Phone Contact: <sip:500@192.168.1.109:5060> Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 369 v=0 o=500 2146032705 0 IN IP4 192.168.1.109 s=SIPPhone Session i=Audio Session c=IN IP4 0.0.0.0 t=0 0 m=audio 16384 RTP/AVP 18 8 0 18 4 9 101 a=sendonly a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (13 headers 17 lines) --- Sending to 192.168.1.109 : 5060 (NAT) <--- Transmitting (NAT) to 192.168.1.109:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.109:5060 ;branch=z9hG4bK-81eef-1fb8d69c-75bdf605;received=192.168.1.109 From: <sip:500@192.168.1.109:5060 >;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8 To: "200"<sip:200@192.168.1.10>;tag=as6b718821 Call-ID: 1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10 CSeq: 1 INVITE Server: FPBX-2.8.1(1.6.2.19) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:200@192.168.1.10> Content-Length: 0 <------------> Audio is at 192.168.1.10 port 18380 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.109:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.109:5060 ;branch=z9hG4bK-81eef-1fb8d69c-75bdf605;received=192.168.1.109 From: <sip:500@192.168.1.109:5060 >;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8 To: "200"<sip:200@192.168.1.10>;tag=as6b718821 Call-ID: 1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10 CSeq: 1 INVITE Server: FPBX-2.8.1(1.6.2.19) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:200@192.168.1.10> Content-Type: application/sdp Content-Length: 235 v=0 o=root 656809389 656809389 IN IP4 192.168.1.10 s=Asterisk PBX 1.6.2.19 c=IN IP4 192.168.1.10 t=0 0 m=audio 18380 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.1.109:5060 ---> ACK sip:200@192.168.1.10 SIP/2.0 From: <sip:500@192.168.1.109:5060 >;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8 To: "200"<sip:200@192.168.1.10>;tag=as6b718821 Call-ID: 1c2ab00b31d6e0c507fc0c94637c88d1@192.168.1.10 CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-81eef-1fb8d6e2-346b6e2d Max-Forwards: 70 User-Agent: SIP Phone Contact: <sip:500@192.168.1.109:5060> Content-Length: 0 The SIP peer is set to canreinvite (if it matters). Does anyone know why it doesn't start the MOH process on this system, unlike the other one? Thanks, Michael
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