Hi all,
I’m facing a strange behaviour with my AsteriskNow 2.10.0rc1(1.8.11) and Genband/Nortel C20 Softswitch.
When I place a call to a Nortel SS and the called number is busy it sends back a SIP 183 message with sendonly to play the busy tone ( in band ).
In this call scenario, AsteriskNow seems to interpret the sendonly present in 183 as HOLD, and so it play the embedded MOH to the asterisk calling user.
Is there a way to configure AsteriskNow no to interpret sendonly like hold ?
This is the relevant part of the sip debug captured with AsteriskNow
<— SIP read from UDP:10.10.10.10:5060 —>
SIP/2.0 183 Session Description
Via: SIP/2.0/UDP 10.20.20.20:5060;received=10.20.20.20;branch=z9hG4bK14184f3c;rport=32699
From: “50289081219” sip:50289081219@test.ip.it;tag=as46810d90
To: sip:0289089584@test.ip.it;tag=SDkihu899-85048
Call-ID: 5b78a6c702eb63037c50c8c23c4126cc@test.ip.it
CSeq: 103 INVITE
Content-Type: application/sdp
Contact: sip:0289089584@10.10.10.10:5060;maddr=10.10.10.10;transport=udp
User-Agent: Nortel SESM 14.0.9.6
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption
Content-Length: 233
v=0
o=NNMAS 1 864 IN IP4 10.10.10.10
s=AFPBX-
e=unknown@invalid.net
c=IN IP4 10.10.10.10
t=0 0
m=audio 34150 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendonly
a=ptime:20
<------------->
— (11 headers 12 lines) —
list_route: hop: sip:0289089584@10.10.10.10:5060;maddr=10.10.10.10;transport=udp
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.10.10:34150
– Call on SIP/50289081219-00000011 placed on hold
– Started music on hold, class ‘default’, on SIP/250-00000010
– SIP/50289081219-00000011 is making progress passing it to SIP/250-00000010
Audio is at 10328
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (no NAT) to 192.168.20.58:5062 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.20.58:5062;branch=z9hG4bK43e92e02f5837e001;received=192.168.20.58
From: “250” sip:250@192.168.20.105:5060;tag=60391607da
To: sip:0289089584@192.168.20.105;tag=as15835675
Call-ID: 918e2f896a980d2d
CSeq: 1580684441 INVITE
Server: AFPBX-2.10.0rc1(1.8.11)-
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:0289089584@192.168.20.105:5060
Content-Type: application/sdp
Content-Length: 229
v=0
o=root 1526534939 1526534939 IN IP4 192.168.20.105
s=AFPBX-
c=IN IP4 192.168.20.105
t=0 0
m=audio 10328 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
Thanks to all for all the help you can give me
TocToc