Sip trunk problem on cme 3825 but works on AsteriksWin32 server

Unanswered Question
Oct 9th, 2009


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 (

- the sip server recive a 53197010 as calling number.

this is mi configuration:

Router#show run

Building configuration...


voice service voip




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


interface GigabitEthernet0/1

description LAN_SOFTPHONE

ip address


ip route


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



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.


INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP;branch=z9hG4bK1B37F

Remote-Party-ID: <sip:[email protected]>;party=calling;screen=yes;privacy=of


From: <sip:[email protected]>;tag=3A9BCB4-17CA

To: <sip:[email protected]>

Date: Wed, 07 Oct 2009 22:12:26 GMT

Call-ID: [email protected]

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 970522294-2999259614-2162464803-3960326021

User-Agent: Cisco-SIPGateway/IOS-12.x



CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1254953546

Contact: <sip:[email protected]: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 ):

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):

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

I have this problem too.
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