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Replies

Help for Seting ut a trunk SIP

florentgoue
Level 1
Level 1

Hi 

Can someone Help me to setup a trunk SIP to my ITSP?

I have a ISR 4321

my router is able to normally ping the ITSP server
But when I look at the status, the connection is not active.

 

Router#show sip-ua register status

Line                       peer           expires(sec)         reg        survival    P-Associ-URI

================================ ========= ============ === ======== ============

+598450             -1                89                        no      normal

 

###########################################################################

 

sip-ua
credentials username +598450 password 6 cHObdaVWSKhiVQI`J\Y[MJgIcEOL[bVNFP[T^cb realm business.com
authentication username +598450@business.com password 6 USIJZaT\`R\TWb_VGN^hZaOBUSWQYb[HNZSAWcY realm business.com
retry invite 2
registrar ipv4:10.203.5.137 expires 60
sip-server ipv4:10.203.5.137
host-registrar

################################################################################

routerI#ping 10.203.5.137
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 10.203.5.137, timeout is 2 seconds:
!!!!!
Success rate is 100 percent (5/5), round-trip min/avg/max = 3/6/16 ms

 

 

13 Replies 13

Mike_Brezicky
Cisco Employee
Cisco Employee
Does the ISP support registration?

Many of my SIP providers do not support registration, but when a call is placed, an authentication challenge is presented (I think SIP 407), which the authentication credentials are then passed.

What happens when you try to place a call to this destination - can you set debug ccsip messages and post the output?

Hi,

What is the configuration under voice service voip? What are you using as
bind interface. You need to choose an interface which will be accepted by
your ITSP. If that is done, then share the output of debug ccsip messages
to see why registration isn't working.

HI Mohammed
Yhx for your reply
Can you tell me hox to choose this interface ??

I'd strongly suggest you review some of the Cisco Live sessions on deploying SIP trunks/CUBE where they go into great detail on best practices and configuration.

HTH

java

if this helps, please rate

HI Jaime
Thx for your reply
please check my last reply

Thx

Thx Jaime
Can you give to me some links

Thx

Go to the on-demand library on Cisco Live site and search for CUBE to find a lot of sessions on the topic, I recommend watching the latest ones.

HTH

java

if this helps, please rate

HI
The provider only need:

Username: +598450
Authuser: +598450@business.com
password: business.com

His sip server addess: 10.203.5.137

thanks

You need to define the bind interface which will be used as source IP for
your register messages. This IP should be the one which your ITSP expect
otherwise they will drop the messages.

I agree with the guys to go through some documentation.

HI Jaime
Thx for your reply
I use the advice of everyone to advance in the config of my trunk.
i now need to remove the port at the register level of my SIP trunk.
when the port is preent, the ITSP refuses the connection.
How to remove this port (in red)?


REGISTER sip:business.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK17695
From: <sip:59840@business.com>;tag=2D047A-B67
To: <sip:59840@business.com>
Date: Wed, 19 Jun 2019 19:31:49 GMT
Call-ID: 350D281A-FFFFFFFF91F911E9-FFFFFFFF8002833A-FFFFFFFFF9DDF090
User-Agent: Cisco-SIPGateway/IOS-16.11.1a
Max-Forwards: 70
Timestamp: 1560972709
CSeq: 25 REGISTER
Contact: <sip:59840@192.168.1.254:5060>
Expires: 60
Supported: path
Content-Length: 0

Hi Mohammed

this is it

voice service voip
ip address trusted list
ipv4 10.203.0.0 255.255.0.0
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 600 min 60

Hello,

I would consider Jaime suggestion on reviewing the documentation.  It looks like you are missing a lot of config information.   I do not see any bind commands under your SIP config.   For the registration piece I have done it this way in the past:

voice class uri "carrier" sip

     host ipv4:"input carrier sip server IP"

     user-id "user id from carrier"

 

A voice class sip profile(s) may also need to be created.  

 

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