Sip trunk configuration cisco router 2811

Unanswered Question
Oct 4th, 2009

Hi, i need help to configure a sip trunk

i have a cisco 2811 with CME and recently bought a sip trunk with a local provider, but i cant get it works.. pls help

the provider said that The sip trunk dont have security, just this parameters to configure.

dir ip X.X.X.X Port 5060

so this is my configuration in the cisco router:

voice service voip

allow-connections sip to sip

sip

registrar server

!

!

voice class codec 1

codec preference 1 g729br8

codec preference 2 g729r8

codec preference 3 g723ar63

codec preference 4 g711ulaw

codec preference 5 g711alaw

!

!

dial-peer voice 11 voip

destination-pattern T

voice-class codec 1

session protocol sipv2

session target ipv4:X.X.X.X

dtmf-relay rtp-nte

no vad

!

!

!

But when a called o recive calls, just hear the busy tone...

and the router dont show any activity

with the debug ccsip all

so pls, someone, what else have to do to configure ? something missing ?

I have this problem too.
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gogasca Mon, 10/05/2009 - 00:30

do they give you any username/password to register?

Normally you use:

sip-ua

authentication username cisco123 password cisco123

Also:

credentials username jose password 12345 realm cisco1.com

credentials username seb password 34567 realm cisco2.net

andresfmg Mon, 10/05/2009 - 13:30

they just give me a ip address x.x.x.x that is a open serv, and a port 5060.

Im using a gateway 2811,

pls help!!!

any comands etc ...anything!!

Dennis Fogler Mon, 10/05/2009 - 16:48

If your inbound calls are getting a fast busy on a SIP trunk it sounds like you might have a codec miss match.

I saw an outbound dial peer but not one to match inbound calls. With dial peers there are two legs per call the ingress and the egress. It looks like you might be hitting the default or dial-peer 0 on the inbound call leg.

dial-peer 0 has VAD on and a default codec of 729r8. If the SIP carrier is not expecting these then you will get a fast busy. It's best to configure an inbound dial-peer using the incoming called-number and the patterns that can be reached internally.

Latly check the number of digits the carrier is sending and what you have defined on your IP Phones. You might need to use a translation pattern to remove digits to match your dial plan.

andresfmg Mon, 10/05/2009 - 19:04

hi thanks for the reply, when i try to recive a call nothings happenss, any activity show the debug ccsip all,

i dont hear any tone...

but when i try to make a call, i hear a busy tone, and show activity in the debug.

so i dont know! what to do!! all papers and pages show the sameconfiguration i had in my router, i dont dont what else to do!!!

I will try a incomin dial-peer, any other idea ?

kelvin.blair Tue, 10/06/2009 - 05:39

I've configured alot of sip trunks and here are some of the issues that an individual typical has while doing the setups.

1.) Codec MisMatch. Like the individual said before, you need to configure an inbound dialpeer to match the called number and allow for it to set the codec type.

2.) Username/password is incorrect or not entered.. The password entered into the configuration on the router is incorrect. So when a call is placed to the provider, the call fails. Check that the provider doesn't require digest authenication.

3.) CLID - Typically the provider is expecting 10 digits to be sent out from the CPE. These 10 digits must match the DID's assigned to the trunk. If a mismatch exist, then the call is dropped with typically error 404. Look at the "debug sip messages" & "debub sip calls" to see what error you are getting.

andresfmg Tue, 10/06/2009 - 06:55

1)i make a voice class codec 1

codec preference 1 g729br8

codec preference 2 g729r8

codec preference 3 g723ar63

codec preference 4 g711ulaw

codec preference 5 g711alaw

2)the trunk dont have username or password, is authentication with ip address.

3)this is the debug , pls help

RouterVoz#

*Oct 6 14:24:37.270: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:3592867@x.x.x.x:5060 SIP/2.0

Via: SIP/2.0/UDP 172.22.24.46:5060;branch=z9hG4bKC2175E

Remote-Party-ID: ;party=calling;screen=yes;privacy=of

f

From: ;tag=1D356354-2578

To:

Date: Tue, 06 Oct 2009 14:24:37 GMT

Call-ID: BCD37594-B1BA11DE-8197E852-24BF8187@172.22.24.46

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

Min-SE: 1800

Cisco-Guid: 2149154573-1213444524-973093377-2887332471

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: 1254839077

Contact:

Expires: 180

Allow-Events: telephone-event

Content-Length: 0

*Oct 6 14:25:09.270: //310/80197F0D3A00/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x7171D6FC

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 : 200.13.230.38:5060

Destn SIP Resp Addr:Port : 200.13.230.38:5060

Destination Name : x.x.x.x

*Oct 6 14:25:09.270: //310/80197F0D3A00/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC) : 16

Disconnect Cause (SIP) : 200

Dennis Fogler Tue, 10/06/2009 - 07:44

Looks like you ran the debug for ccsip calls and it looks like the call was cleared normally. Also might want to look at the following debugs independently.

1) debug ccsip messages (mentioned above)

2) debug ccsip events

3) debig ccsip errors.

andresfmg Tue, 10/06/2009 - 08:43

This debugs is for outbounds calls,the incomming calls dont show anything in the debug....

(note: i changed the ip of the sip provider to x.x.x.x)

1)*Oct 6 16:33:55.549: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:3592867@x.x.x.x:5060 SIP/2.0

Via: SIP/2.0/UDP 172.22.24.46:5060;branch=z9hG4bKC461C

Remote-Party-ID: ;party=calling;screen=yes;privacy=of

f

From: ;tag=1DABC51C-1D14

To:

Date: Tue, 06 Oct 2009 16:33:55 GMT

Call-ID: CD1EF6A0-B1CC11DE-819DE852-24BF8187@172.22.24.46

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

Min-SE: 1800

Cisco-Guid: 9123870-2540810668-16792577-2887332471

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: 1254846835

Contact:

Expires: 180

Allow-Events: telephone-event

Content-Length: 0

2)

*Oct 6 16:36:53.465: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event

from SIP SPI : SIPSPI_EV_CC_CALL_SETUP

*Oct 6 16:36:53.469: //316/8069CB9A0200/SIP/Event/sipSPICreateRpid: Received Oc

tet3A=0x81 -> Setting ;screen=yes ;privacy=off

*Oct 6 16:36:53.969: //316/8069CB9A0200/SIP/Event/sipSPICreateRpid: Received Oc

tet3A=0x81 -> Setting ;screen=yes ;privacy=off

*Oct 6 16:36:54.969: //316/8069CB9A0200/SIP/Event/sipSPICreateRpid: Received Oc

tet3A=0x81 -> Setting ;screen=yes ;privacy=off

*Oct 6 16:36:56.969: //316/8069CB9A0200/SIP/Event/sipSPICreateRpid: Received Oc

tet3A=0x81 -> Setting ;screen=yes ;privacy=off

*Oct 6 16:37:00.969: //316/8069CB9A0200/SIP/Event/sipSPICreateRpid: Received Oc

tet3A=0x81 -> Setting ;screen=yes ;privacy=off

*Oct 6 16:37:08.969: //316/8069CB9A0200/SIP/Event/sipSPICreateRpid: Received Oc

tet3A=0x81 -> Setting ;screen=yes ;privacy=off

*Oct 6 16:37:24.969: //316/8069CB9A0200/SIP/Event/sipSPICreateRpid: Received Oc

tet3A=0x81 -> Setting ;screen=yes ;privacy=off

*Oct 6 16:37:53.485: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event

from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT

3)*Oct 6 16:40:40.401: //318/003DF3FB0300/SIP/Error/act_sentinvite_wait_100: Out

of retries

andresfmg Thu, 10/08/2009 - 17:12

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: ;party=calling;screen=yes;privacy=of

f

From: ;tag=3A9BCB4-17CA

To:

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:

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

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Posted October 4, 2009 at 1:37 PM
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