Hi,
we have a customer with some weird issuess on sip level.
They have an Aastra5000 gateway that has a peer with our Asterisk server to call and receive external lines.
Problem is only on outgoing communications. Sometimes, the customer receives a Busy Here 486 for a short period of time when trying to call to the external. This is the flow we get when the problem occurs.
Invite from Aastra to our Asterisk
Trying from our Asterisk to the Aastra
Immediately after this our Asterisk sends out a Busy Here even when the called party is NOT busy.
This is a special configuration-> we use nat on our trunk because off the customer’s SDP is a private unknown address to our Asterisk. Asterisk is 1.3.5.8 and Aastra5000 is 1.135.7.61(Behind Firewall).
INVITE sip:015630392@1.3.5.8:5060 SIP/2.0
Via: SIP/2.0/UDP 1.135.7.61:65476;ctxe=00003188;branch=z9hG4bK_1_INVITE_0201FFFF3CFE1181_A5000;rport
From: sip:3214811141@1.135.7.61:65476;tag=C301EE7E_nab_FFFF_isp_FFFF_mgt_4179
To: sip:015630392@1.3.5.8:5060
Call-ID: 0201FFFF3CFE1181
CSeq: 1 INVITE
Contact: sip:3214811141@1.135.7.61:65476;ctxe=00003188
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE
User-To-User: 04;encoding=hex;purpose=isdn-interwork;content=isdn-uui
Max-Forwards: 70
Remote-Party-ID: sip:3214811141@192.168.0.200;party=calling;privacy=off
Proxy-Require: privacy
User-Agent: A5000 R5.3 SP4 /C502 BEL
Content-Type: application/sdp
Content-Length: 236
v=0
o=- 0 0 IN IP4 192.168.0.201
s=-
c=IN IP4 192.168.0.201
t=0 0
m=audio 40002 RTP/AVP 8 0 101
a=rtcp:40003
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=ptime:30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.135.7.61:65476;ctxe=00003188;branch=z9hG4bK_1_INVITE_0201FFFF3CFE1181_A5000;received=1.135.7.61;rport=65476
From: sip:3214811141@1.135.7.61:65476;tag=C301EE7E_nab_FFFF_isp_FFFF_mgt_4179
To: sip:015630392@1.3.5.8:5060
Call-ID: 0201FFFF3CFE1181
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.15-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:015630392@1.3.5.8:5060
Content-Length: 0
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 1.135.7.61:65476;ctxe=00003188;branch=z9hG4bK_1_INVITE_0201FFFF3CFE1181_A5000;received=1.135.7.61;rport=65476
From: sip:3214811141@1.135.7.61:65476;tag=C301EE7E_nab_FFFF_isp_FFFF_mgt_4179
To: sip:015630392@1.3.5.8:5060;tag=as68d8f52d
Call-ID: 0201FFFF3CFE1181
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.15-cert3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Reason: Q.850;cause=17
Content-Length: 0
ACK sip:015630392@1.3.5.8:5060 SIP/2.0
Via: SIP/2.0/UDP 1.135.7.61:65476;ctxe=00003188;branch=z9hG4bK_1_INVITE_0201FFFF3CFE1181_A5000;rport
From: sip:3214811141@1.135.7.61:65476;tag=C301EE7E_nab_FFFF_isp_FFFF_mgt_4179
To: sip:015630392@1.3.5.8:5060;tag=as68d8f52d
Call-ID: 0201FFFF3CFE1181
CSeq: 1 ACK
Contact: sip:3214811141@1.135.7.61:65476;ctxe=00003188
Max-Forwards: 70
User-Agent: A5000 R5.3 SP4 /C502 BEL
Content-Length: 0
I’m searching in the early media domain or the natting we configure on our trunk to send back rtp towards 1.135.7.61 in stead of 192.168.0.201.
Do you guys have any idea here ? thx
Jan Meylaers