[Asterisk-Users] VoIP network configuration using Asterisk and SIP
Moises Silva
moises.silva at gmail.com
Tue Apr 5 06:54:34 MST 2005
Hi. I have no time to read the whole configuration of each device, but
first, i guess you have to be sure that each PBX is able to reach the
other networks, may be a simple ping can tell you this. After that, i
think you have to make sure that no firewall is blocking the 5060 port
for the SIP calls. An finally, could you please post the Asterisk
console output ??
Also you can try to make a direct IP-IP-CALL, is pretty much likely
that your phones allow you to do that, try it to see what happend.
Good Look.
- Moy
On Apr 5, 2005 1:32 PM, Mariña Varela Senín <marinhavs at gmail.com> wrote:
> 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.
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