[asterisk-users] a=sendonly Music On Hold ignored

Tue Jul 19 02:20:47 CDT 2011

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 at 10.0.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;rport;branch=z9hG4bK337477455
From: "1001" <sip:1001 at 10.0.0.10>;tag=446928907
To: <sip:21 at 10.0.0.10>;tag=as479a82ac
Call-ID: 65159842 at 10.0.0.2
CSeq: 22 INVITE
Contact: <sip:1001 at 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 at 10.0.0.10>;tag=446928907
To: <sip:21 at 10.0.0.10>;tag=as479a82ac
Call-ID: 65159842 at 10.0.0.2
CSeq: 22 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:21 at 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 at 10.0.0.10>;tag=446928907
To: <sip:21 at 10.0.0.10>;tag=as479a82ac
Call-ID: 65159842 at 10.0.0.2
CSeq: 22 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:21 at 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 at 192.168.1.10 SIP/2.0
From: <sip:500 at 192.168.1.109:5060
>;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8
To: "200"<sip:200 at 192.168.1.10>;tag=as6b718821
Call-ID: 1c2ab00b31d6e0c507fc0c94637c88d1 at 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 at 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 at 192.168.1.109:5060
>;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8
To: "200"<sip:200 at 192.168.1.10>;tag=as6b718821
Call-ID: 1c2ab00b31d6e0c507fc0c94637c88d1 at 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 at 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 at 192.168.1.109:5060
>;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8
To: "200"<sip:200 at 192.168.1.10>;tag=as6b718821
Call-ID: 1c2ab00b31d6e0c507fc0c94637c88d1 at 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 at 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 at 192.168.1.10 SIP/2.0
From: <sip:500 at 192.168.1.109:5060
>;tag=80d12d18-c0a8016d-13c4-45026-81ee8-4c3fc8ef-81ee8
To: "200"<sip:200 at 192.168.1.10>;tag=as6b718821
Call-ID: 1c2ab00b31d6e0c507fc0c94637c88d1 at 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 at 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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