[Asterisk-Users] VoIP network configuration using Asterisk and SIP

Tue Apr 5 06:32:24 MST 2005

Hallo,

I am trying to configure a VoIP network using two routers Cisco 2600,
each one connected to an Asterisk PBX; there is also one softelephone
connected to each PBX.

----> Figure:
--------------           -----------           ---------          
---------       ------------            -------------
softphone ------- asterisk A ------- router A ----- router B ----
asterisk B ------- softphone
--------------           -----------           ---------          
---------        ------------           -------------

----> Interfaces:
AsteriskA ---> fastEthernet ---> RouterA
RouterA ---> E1 ----> RouterB
RouterB --> fastEthernet --> AsteriskB

----> O.S:

Asterisk A runs on Linux Ubuntu
Asterisk B runs on Windows XP

----> IP addresses:

AsteriskA (192.168.1.2) --- (192.168.1.1) Router A
Router A (192.168.2.1) --- (192.168.2.2) Router B
Router B (192.168.3.1) --- (192.168.3.2) Asterisk B

----> Softphones

In Linux we use a Kphone
In Windows we use X-lite

----> Configuration of the Routers.

_________________________

Router A: 
_________________________

hostname RouterA
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$JPk1$rv8t.miyvT6VlxmGYUcIw0
!
clock timezone GMT 0
no network-clock-participate slot 1
no network-clock-participate wic 0
no aaa new-model
ip subnet-zero
ip cef
!
!
no ftp-server write-enable
!
!
voice service voip
 sip
  bind all source-interface FastEthernet0/0
!
!
controller E1 0/0
 channel-group 0 timeslots 1-31
!
!
interface FastEthernet0/0
 ip address 192.168.1.1 255.255.255.0
 ip rip send version 1 2
 ip rip receive version 1 2
 duplex auto
 speed auto
!
interface Serial0/0:0
 ip address 192.168.2.1 255.255.255.0
 ip rip send version 1 2
 ip rip receive version 1 2
 encapsulation ppp
 ip tcp header-compression iphc-format
 fair-queue 64 256 47
 ip rtp header-compression iphc-format
 ip rtp compression-connections 30
 ip rsvp bandwidth 1488 64
!
router rip
 version 2
 network 192.168.1.0
 network 192.168.2.0
 network 192.168.3.0
!
ip classless
ip http server
!
!
dial-peer voice 1 voip
 destination-pattern 06815678
 session protocol sipv2
 session target ipv4:192.168.2.2
 session transport udp
 codec g711ulaw
!
sip-ua
 sip-server ipv4:192.168.1.2
!

_________________________

Router B: 
_________________________

service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname RouterB
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$VpOE$LTlD1CJRtmaLA9uBnS6Z0.
!
clock timezone GMT 0
no network-clock-participate slot 1
no network-clock-participate wic 0
no aaa new-model
ip subnet-zero
ip cef
!
!
no ftp-server write-enable
!
!
voice service voip
 sip
  bind all source-interface FastEthernet0/0
!
!
controller E1 0/0
 channel-group 0 timeslots 1-31
!
!
interface FastEthernet0/0
 ip address 192.168.3.1 255.255.255.0
 ip rip send version 1 2
 ip rip receive version 1 2
 duplex auto
 speed auto
!
interface Serial0/0:0
 ip address 192.168.2.2 255.255.255.0
 ip rip send version 1 2
 ip rip receive version 1 2
 encapsulation ppp
 ip tcp header-compression iphc-format
 fair-queue 64 256 47
 ip rtp header-compression iphc-format
 ip rtp compression-connections 30
 ip rsvp bandwidth 1488 64
!
router rip
 version 2
 network 192.168.1.0
 network 192.168.2.0
 network 192.168.3.0
!
ip classless
ip http server
!
!
dial-peer voice 1 voip
 destination-pattern 06811234
 session protocol sipv2
 session target ipv4:192.168.2.1
 session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
sip-ua
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 timers trying 1000
 sip-server ipv4:192.168.3.2
!

----> Configuration of the Asterisk

________________________

Asterisk A:
________________________

-- sip.conf

[general]
context = default
port = 5060
binaddr = 0.0.0.0
disallow = all
allow = ulaw
maxexpirey = 1500
defaultexpirey = 160
nat = no

[kphone]
type = friend
context = local-phone
username = kphone
host = dynamic
dtmfmode = inband
nat = no
disallow = all
allow = ulaw
callerid = "ubuntu" <06811234>

[192.168.1.1]
context = incoming
type = friend
host = 192.168.1.1
dtmfmode = rfc2833
disallow = all
allow = ulaw
callerid = "externa" <06815678>

-- extensions.conf

[general]
static = yes
writeprotect = no

[default]
exten => _.,1,Congestion

[incoming]
include => lan-phones

[local-phones]
include => lan-phones
include => outbound

[outbound]
exten => 06815678,1,Dial(SIP/${EXTEN}@192.168.1.1)
exten => 06815678,2,Congestion

[lan-phones]
exten => 06811234,1,Wait(2)
exten => 06811234,2,Playback(vm-goodbye)
exten => 06811234,3,Hangup

________________________

Asterisk B:
________________________

-- sip.conf

[general]
context = default
port = 5060
binaddr = 0.0.0.0
disallow = all
allow = ulaw
nat = no

[xlite]
type = friend
context = local-phone
username = xlite
host = dynamic
dtmfmode = inband
nat = no
careinvite = no
disallow = all
allow = ulaw

[192.168.3.1]
context = incoming
type = friend
host = 192.168.3.1
dtmfmode = rfc2833
disallow = all
allow = ulaw

-- extensions.conf

[general]
static = yes
writeprotect = no

[default]
exten => _.,1,Congestion

[incoming]
include => lan-phones

[local-phones]
include => lan-phones
include => outbound

[outbound]
exten => 06811234,1,Dial(SIP/${EXTEN}@192.168.3.1)
exten => 06811234,2,Congestion

[lan-phones]
exten => 06815678,1,Wait(2)
exten => 06815678,2,Playback(vm-goodbye)
exten => 06815678,3,Hangup

********************************************************************************************
The problem is the following:

- All internal calls, go
- When we try to call from one point to the network to the another it doesn't go

For example, we try to call from the Kphone of linux, to the X-lite of Asterisk

We dial: 06815678 and it doesn't go

If we debug in Router A the CCSIP information we have:

-----------------------------------------------------------
Mar  1 02:14:18.822: CCSIP-SPI-CONTROL:  sipSPIMatchSrcIpGroup: Match not found
 on carrier id
*Mar  1 02:14:18.822: CCSIP-SPI-CONTROL:  sipSPIMatchSrcIpGroup: Match
not found on Incoming called number: 06815678
*Mar  1 02:14:18.826: CCSIP-SPI-CONTROL:  sipSPIMatchSrcIpGroup: Match
not found on destination pattern: 06811234
.
.
.
.
.
.
.
*Mar  1 02:15:03.207:
 Disconnect Cause (CC)    : 3
Disconnect Cause (SIP)   : 404
-----------------------------------------------------------

I have been working a whole week with these, but I don't find the
mistake. If anyone could help me, I'll be very grateful.

Thanks in advance.