10-04-2009 01:37 PM - edited 03-18-2019 10:43 AM
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 ?
10-05-2009 12:30 AM
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
10-05-2009 01:30 PM
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!!
10-05-2009 04:48 PM
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.
10-05-2009 07:04 PM
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 ?
10-06-2009 05:39 AM
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.
10-06-2009 06:55 AM
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: <53197010>;party=calling;screen=yes;privacy=of53197010>
f
From: <53197010>;tag=1D356354-257853197010>
To: <3592867>3592867>
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: <53197010>53197010>
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
10-06-2009 07:44 AM
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.
10-06-2009 08:43 AM
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: <53197010>;party=calling;screen=yes;privacy=of53197010>
f
From: <53197010>;tag=1DABC51C-1D1453197010>
To: <3592867>3592867>
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: <53197010>53197010>
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
10-06-2009 08:47 AM
Is there a firewall between thi router and the SIP Provider?
10-06-2009 08:48 AM
no, its clear...
10-08-2009 05:12 PM
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: <53197010>;party=calling;screen=yes;privacy=of53197010>
f
From: <53197010>;tag=3A9BCB4-17CA53197010>
To: <3592867>3592867>
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: <53197010>53197010>
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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