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SIP TRUNK BETWEEN TRIXBOX AND CME

Jesutofunmi O
Level 1
Level 1

Hello Guys,

 

I have a Cisco CME with about 40 phones on LAN (site 1) which can call each other properly without any issues (over the LAN network). I have a functional VPN to a remote site. The remote site (site 2) users use Trixbox for their own LAN communication. Now, I wish to establish a SIP trunk between the CME and the Trixbox so both site can call each other. Has anyone done this before? Please assist with a configuration guide. I am not confident of the ones I have seen online and there are very few materials out there for this type of deployment. Please note that I am beginner in SIP. 

1 Accepted Solution

Accepted Solutions

Hello Guys,

Sorry I am posting this late but I got the SIP trunk working already. I will post the config below so other people who visit can find it helpful. What I did on the CME and Trixbox;

1. was to create dial-peers on the CME

2. Create Voice service and "allow sip to sip connections". Remember my Trixbox speaks SIP.

3. Create dial peers for both incoming and outgoing calls for the trixbox.

 

4. On the trixbox, I configured the trunk (which includes outgoing and incoming call settings), filled the required parameters

5. Created an outbound route, and called up the trunk I had created from the outbound route settings.

N.B: please note that I had configured an L2L VPN between the two sites and intial configuration of the CME had been done such that the phones can make LAN calls.

  

Create an outbound route

 

View solution in original post

20 Replies 20

R0g22
Cisco Employee
Cisco Employee
If you know how to create dial-peers, then all you need is a SIP VoIP dial-peer pointing to the Trixbox.

Dennis Mink
VIP Alumni
VIP Alumni

on your cme point a dial-peer with the DN pattern of your remote site for example 1...  and use sip on the dial peer. point trixbox to your cme in the same sort of fashion.

Please remember to rate useful posts, by clicking on the stars below.

Alright, thanks. I'll do just that.

Hello Guys,

 

I have been able to configure the CME to call Trixbox successfully but Trixbox is unable to call CME. Also, I intend to trunk from CME to other boxes but I am unable to create multiple SIP registrars as I keep getting the error below when I try it 

POHT (config-sip-ua)#registrar 1 ipv4:192.168.2.25 expires 3600
Primary-Secondary mode currently configured; Cannot configure multiple registrar

 

 

This is my CME Config

 

sh run
Building configuration...


version 15.3
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname POHT
!
boot-start-marker
boot-end-marker
!
aqm-register-fnf
!
! card type command needed for slot/vwic-slot 0/1
logging buffered 51200 warnings
!
no aaa new-model

!
!
!
ip domain name bwl.com
ip cef
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
!
crypto pki trustpoint TP-self-signed-2743843390
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2743843390
revocation-check none
rsakeypair TP-self-signed-2743843390
!

voice-card 0
dspfarm
dsp services dspfarm
!
!
!
voice service voip
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 3600 min 600
no call service stop
!
!
voice register global
mode cme
source-address 172.16.121.5 port 5060
max-dn 200
max-pool 165
authenticate realm

!
!
!
!
!
license udi pid
hw-module ism 0
!
hw-module pvdm 0/0
!
hw-module pvdm 0/1
!
!
!
username admin privilege 15 secret 
!
redundancy
!
!
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
no ip address
shutdown
duplex auto
speed auto
!
interface ISM0/0
no ip address
shutdown
!Application: CUE Running on ISM
!
interface GigabitEthernet0/1
description sw
ip address 172.16.121.5 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
!
interface ISM0/1
description Internal switch interface connected to Internal Service Module
no ip address
!
interface Vlan1
no ip address
!
ip forward-protocol nd
!
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:GUI
!
ip route 0.0.0.0 0.0.0.0 172.16.121.1
!
!
nls resp-timeout 1
cpd cr-id 1
!
tftp-server flash:/term45.default.loads alias term45.default.loads
!
control-plane
!
!
voice-port 0/2/0
!
voice-port 0/2/1
!
voice-port 0/2/2
!
voice-port 0/2/3
!
voice-port 0/3/0
!
voice-port 0/3/1
!
voice-port 0/3/2
!
voice-port 0/3/3
!
!
!
!
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local GigabitEthernet0/1
sccp ccm 172.16.121.5 identifier 1 priority 1 version 7.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/1
associate ccm 1 priority 1
associate profile 1 register POHT-CONF-BRDG
!
dspfarm profile 1 conference
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 6
associate application SCCP

!
dial-peer voice 1 voip
destination-pattern 1..
session protocol sipv2
session target ipv4:192.168.0.120
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 2 voip
destination-pattern 2..
session protocol sipv2
session target ipv4:192.168.2.25
dtmf-relay sip-notify
codec g711ulaw
no vad
!
!
sip-ua
registrar ipv4:192.168.0.120 expires 3600
sip-server ipv4:192.168.0.120
!
!
!
gatekeeper
shutdown
!
!
telephony-service
sdspfarm units 10
sdspfarm tag 1 POUT
conference hardware
max-ephones 165
max-dn 200

 

 

TRIXBOX 1: 192.168..0.120

TRIXBOX 2: 192.168.2.25

1. You can configure upto 6 registrars. You need to change the index, you already have a registrar for index 1.
2. Collect logs for a failed call -
debug ccsip message
debug ccsip error
debug voice ccapi inout

Hi Nipun,

 

Thanks for your response. 

1. I removed the registrar and I could still call the trixbox only with the dial peer. But the trixbox is still unable to call CME. When I try to add more registrars, I get the error message below;

*Primary-Secondary mode currently configured; Cannot configure multiple registrar*

Please, how can I place the CME in a mode to accept more than 1 registrar?

 

2. Does this mean I cannot add multiple dial peers (multiple trixboxes) without adding multiple registrars?

And if I may ask please, what is the difference between the registrar server and a register server?

 

Share output for "show run | s sip-ua"

Registrars are used for SIP end points registrations. With dial-peers you govern call routing. You only need registrar if you are trying to register something to the registrar IP in this case would be your CME. Does your trixbox require this ? If not, you don't need registrar.

Nipun,

 

Below please;

#show run | s sip-ua
sip-ua
registrar ipv4:192.168.0.120 expires 3600
sip-server ipv4:192.168.0.120

 

 

 I get the error message below when i try to create multiple registrars

*Primary-Secondary mode currently configured; Cannot configure multiple registrar*

Like I mentioned in my other post, you need to index the registrar servers. Something like -

registrar 1 ipv4:1.1.1.1
registrar 2 ipv4:2.2.2.2

You configure upto 6 Registrars but you need to understand if you need a Registrar or not. Read my last post.

Hi Nipun,

 

I have been able to create multiple registrars but;

1. Still unable to make call to the other peer 192.168.2.25

2. Still unable to make calls to CME from either of the peers 192.168.0.120 and 192.168.2.25

 

Calls from trixbox (192.168.0.120) to CME works just fine. Kindly help look into the error messages received from debug. I made a call to an extension of 192.168.2.25 and took down the error messages. 

I don't see a single call in the logs. Do the following first -

logg buff 5000000 deb
no logg con
no logg mon
no logg trap
no logg source
logg on
do term no mon

Post above, enable the following -

debug ccsip message
debug ccsip error
debug voice ccapi inout

Once you have made the test calls, increase the scrollback buffer of your telnet client then do the following -

term len 0
show logg

Copy the logs in a text file and send across.

Hi Nipun,

Please find attached.

 

Thank you.

What scenario were the logs collected for ? Calls from CME or calls to CME ? I still don't see a call.