Hello,
My Cisco AS5400 is connected to a sip provider to make and receive calls. The AS5400 is connected to a huawei core backbone.
Huawei Core -- Cisco AS5400 --(Internet)-- SIP Provider
Everything is working fine (inbound and outbound calls) except when an inbound call cannot connected due to a busy or not answering phone.
Huawei Core -- Cisco AS5400 --(Internet)-- SIP Provider -- PSTN
GSM not answering <-------(Dial Peer 30) -----CALL-----(Dial Peer 20) -------SIP Provider -- PSTN
When it is the case, I received from the core huawei :
Received:
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.30.1.46:5060;branch=z9hG4bK577F59
Call-ID: 203BF3AB-258011D7-917E83E8-9B382CF3@10.30.1.46
From: <sip:6267381313@10.30.1.46>;tag=FC74AF48-23A3
To: <sip:1398562077827637@10.30.1.10>;tag=b1cade24
CSeq: 101 INVITE
Reason: Q.850;cause=16;text="Normal call clearing"
Content-Length: 0
But this is not forwarded to the sip provider as expected,
Actually, the cisco gateway sent a new invite to the sip provider. Like it is matching the inbound dialpeer as the new outbound dialpeer to place calls.
Huawei Core -- Cisco AS5400 --(Internet)-- SIP Provider -- PSTN
GSM not answering <-------(Dial Peer 30) -----SIP GW-----(Dial Peer 20) -------SIP Provider -- PSTN
|
|
------------- (480)---------> SIP GW---- (INVITE) ---------> SIP Provider
Sent:
INVITE sip:52454236267381313@213.218.126.214:5060 SIP/2.0
Via: SIP/2.0/UDP 201.124.181.156:5060;branch=z9hG4bK57A1F13
Remote-Party-ID: <sip:6267381313@201.124.181.156>;party=calling;screen=no;privacy=off
From: <sip:6267381313@201.124.181.156>;tag=FC7506FC-15F0
To: <sip:52454238562077827637@213.218.126.214>
Date: Sun, 12 Jan 2003 16:17:28 GMT
Call-ID: 2D9D3FB5-258011D7-918083E8-9B382CF3@201.124.181.156
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 540719963-629150167-2440594408-2604150003
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1042388248
Contact: <sip:6267381313@201.124.181.156:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 2586 5727 IN IP4 201.124.181.156
s=SIP Call
c=IN IP4 201.124.181.156 --More-- t=0 0
m=audio 17188 RTP/AVP 18 101 19
c=IN IP4 201.124.181.156
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
voice translation-rule 1
rule 1 // /524/
!
voice translation-rule 2
rule 2 /5423/ /139/
!
!
voice translation-profile 1
translate called 1
!
voice translation-profile 2
translate called 2
!
dial-peer voice 30 voip
description INSIDE
translation-profile outgoing 2
destination-pattern 5423.+
voice-class codec 3
session protocol sipv2
session target ipv4:10.30.1.10
dtmf-relay rtp-nte
!
dial-peer voice 20 voip
description OUTSIDE
translation-profile outgoing 1
destination-pattern .+
voice-class codec 3
session protocol sipv2
session target ipv4:213.218.126.214
dtmf-relay rtp-nte
Why this behaviour ? We need the sip gateway to just forward the 480 Message to the SIP provider.
