10-09-2009 04:59 AM - edited 03-15-2019 05:11 AM
Hi,
I need help.
Im trying to configure a sip trunk on my cme 3825, but i cant get works.
i made a call and the other side ring but thats all. just noise in both sides.
i debug the ccsip messages and i saw that i sent invite messages, but never recived
ack or any message from the sip-server.
The weird thing is that the trunk is tested by the local provider with
a asteriksWin32 Pbx and the calls incoming and recive are just fine!!!
so pls, what wrong with mi router !!!
the provider told the parameters of the sip trunk
- its sip-server A.B.C.D
- its a ip athenticate based (172.22.24.46)
- the sip server recive a 53197010 as calling number.
this is mi configuration:
Router#show run
Building configuration...
!
voice service voip
sip
!
!
voice class codec 1
codec preference 1 g729br8
codec preference 2 g729r8
codec preference 3 g723ar63
codec preference 4 g711ulaw
codec preference 5 g711alaw
!
voice translation-rule 1
rule 1 /^.*/ /53197010/
!
voice translation-profile out5
translate calling 1
!
interface GigabitEthernet0/0
description TRONCAL SIP
ip address 172.22.24.46 255.255.255.252
!
interface GigabitEthernet0/1
description LAN_SOFTPHONE
ip address 172.25.51.252 255.255.254.0
!
ip route 0.0.0.0 0.0.0.0 172.22.24.45
!
dial-peer voice 11 voip
description outgoing sip calls
translation-profile outgoing out5
service session
destination-pattern T
voice-class codec 1
session protocol sipv2
session target ipv4:A.B.C.D
dtmf-relay rtp-nte
clid network-number 53197010
no vad
!
dial-peer voice 200000 voip
description incoming sip calls
voice-class codec 1
session protocol sipv2
incoming called-number T
dtmf-relay sip-notify rtp-nte
!
sip-ua
registrar ipv4:A.B.C.D expires 3600
!
----------------------------------------the debug ccsip messages
mi debug cccsip show that i send sip invite packets but no response from the server openser.
Sent:
INVITE sip:3592867@A.B.C.D:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.24.46:5060;branch=z9hG4bK1B37F
Remote-Party-ID: <sip:53197010@172.22.24.46>;party=calling;screen=yes;privacy=of
f
From: <sip:53197010@172.22.24.46>;tag=3A9BCB4-17CA
To: <sip:3592867@A.B.C.D>
Date: Wed, 07 Oct 2009 22:12:26 GMT
Call-ID: 419E2136-B2C511DE-80E99823-EC0DC785@172.22.24.46
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 970522294-2999259614-2162464803-3960326021
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF
Y, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1254953546
Contact: <sip:53197010@172.22.24.46:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 245
-----------------------------
finally shows...
*Oct 7 22:12:58.199: //74/39D8FEB680E4/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x65EC0340
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 53197010
Called Number : 3592867
Source IP Address (Sig ): 172.22.24.46
Destn SIP Req Addr:Port : A.B.C.D:5060
Destn SIP Resp Addr:Port : A.B.C.D:5060
Destination Name : A.B.C.D
*Oct 7 22:12:58.199: //74/39D8FEB680E4/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 172.22.24.46
Source IP Port (Media): 16446
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Oct 7 22:12:58.199: //74/39D8FEB680E4/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 102
Disconnect Cause (SIP) : 200
10-09-2009 11:56 AM
Are you behind nat? Cisco SIP works only behind NAT done by cisco.
10-13-2009 05:32 AM
This would be the problem:
Contact: <53197010> 53197010>
If like Paolo said and you are behind NAT, you need to have SIP fix-up enabled. As well, sometimes providers will do NAT traversal. You may want to add this command:
sip-ua
connection-reuse
Otherwise, you need a router capable of doing NAT fixup.
-nick
10-13-2009 06:38 AM
sip-ua
connection-reuse
That is hidden and undocumented, so one would need to know a bit more about it :)
10-13-2009 08:48 AM
Normally when a router initiates a SIP request the source port will be a random port above 1024.
This is in contrast to many SIP endpoints which use 5060 as the source and destination port, even though the application only depends on the destination port.
Many SIP providers will automatically contact the source port when they see a NAT address hoping that they can get through the firewall.
It's a long shot, but I've seen it work before.
-nick
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