VG224 Calls within itself

[campbellj1]

Hello

I'm not sure how to describe this, perhaps it's a feature however I'm currently experimenting with a VG224 and have it set up with our Asterisk PBX via SIP.

It's all working perfect however I've just realised that if I have a phone in port 0 and a phone in port 1 when Port 0 dials the extension for Port 1 the call routes within the VG224 rather than going back to the SIP server then back to the VG224.

I can see how handy this would be however we need all calls to go to the PBX for call logging and other features (such as forwarding etc)

Can anyone explain how to configure this?

Thanks

[George Thomas]

Can you post your VG224 config? You probably have to use class of restrictions on the VG224 to send calls back to the SIP server and back to the VG224.

[gousuman · Accepted solution]

Hi Campbellj1

As per my understanding the analog ports on the VG is just acting like a  FXS ports on a voice gateway.
I believe you might be having dial-peers for the two analog ports on the VG , so when the analog phone on 1st port calls to the analog phone on 2nd port , the call comes in through 1st port using the incoming POTS dial-peer and then the call is sent to the 2nd port as per matching the dial-peer pointing to the second analog port.
Hairpinning of call.

Regarding your requirement where you need to send the call to the PBX , we can play around with the dial-peers,
we can use higher preference to the voip dial-peer pointing to the PBX then the pots dial-peer pointing to the 2nd analog port, and so the call would reach pbx .

[Anil Rajan · Accepted solution]

Hi Campbellj1,

Please provide the sh run for checking the dial peer configuration.

Thanks,

Anil

[campbellj1]

version 12.4
no service pad
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
!
hostname VG224
!
boot-start-marker
boot-end-marker
!
logging message-counter syslog
!
no aaa new-model
ip source-route
ip cef
!
!
no ipv6 cef
!
!
!
!
!
voice call send-alert
voice rtp send-recv
!
voice service pots
!
voice service voip
 fax protocol pass-through g711alaw
 modem passthrough nse codec g711alaw
 sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0
!
voice class codec 1
 codec preference 1 g711alaw
!
!
!
!
!
!
!
!
!
!
!
!
!
voice-card 0
!
!
application
 service dsapp
  param callWaiting TRUE
  param callTransfer TRUE
 !
 global
  service default dsapp
 !
!
archive
 log config
  hidekeys
!
!
!
!
!
interface FastEthernet0/0
 ip address 10.10.0.15 255.255.252.0
 duplex auto
 speed auto
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.10.0.10
no ip http server
ip http authentication local
!
!
!
control-plane
!
!
!
voice-port 2/0
 ring cadence pattern11
 alt-battery-feed feed2
 cptone GB
 timeouts call-disconnect 5
 timing hookflash-in 110 70
 impedance complex1
 station-id name Port 0
 station-id number 3013
 caller-id enable
 caller-id alerting line-reversal
!
voice-port 2/1
 ring cadence pattern11
 alt-battery-feed feed2
 cptone GB
 timeouts call-disconnect 5
 timing hookflash-in 110 70
 impedance complex1
 station-id name Port 1
 station-id number 3133
 caller-id enable
 caller-id alerting line-reversal
!
voice-port 2/2
!
voice-port 2/3
!
voice-port 2/4
!
voice-port 2/5
!
voice-port 2/6
!
voice-port 2/7
!
voice-port 2/8
!
voice-port 2/9
!
voice-port 2/10
!
voice-port 2/11
!
voice-port 2/12
!
voice-port 2/13
!
voice-port 2/14
!
voice-port 2/15
!
voice-port 2/16
!
voice-port 2/17
!
voice-port 2/18
!
voice-port 2/19
!
voice-port 2/20
!
voice-port 2/21
!
voice-port 2/22
!
voice-port 2/23
!
ccm-manager fax protocol cisco
!
!
!
!
dial-peer voice 10 pots
 destination-pattern 3133
 port 2/0
 authentication username 3133 password 7
!
dial-peer voice 100 voip
 destination-pattern 3...
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay sip-notify rtp-nte
 no vad
!
dial-peer voice 101 voip
 destination-pattern 5...
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 dtmf-relay sip-notify rtp-nte
 no vad
!
dial-peer voice 102 voip
 preference 1
 destination-pattern 7...
 voice-class codec 1
 session protocol sipv2
 session target ipv4:10.10.0.11
 dtmf-relay sip-notify rtp-nte
 no vad
!
dial-peer voice 11 pots
 destination-pattern 3013
 port 2/1
 authentication username 3013 password 7
!
!
sip-ua
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 timers trying 1000
 mwi-server ipv4:10.10.0.10 expires 3600 port 5060 transport udp
 registrar ipv4:10.10.0.10:5060 expires 3600
 registrar ipv4:10.10.0.11:5060 expires 3600 secondary
 sip-server ipv4:10.10.0.10
 offer call-hold conn-addr
!
!
line con 0
line aux 0
line vty 0 4
 privilege level 15
 logging synchronous
 login local
!
end

[ravmuthu]

We will explore and update you

[Anil Rajan · Accepted solution]

Hi,

If you are making call from port 0 to port 1 by dialing 3013, it is going to take Dial peer 11, which has most number of matched digit (matched digits 4), rather taking DP 100 (matched digits 1).

dial-peer voice 11 pots
 destination-pattern 3013
 port 2/1

dial-peer voice 100 voip
 destination-pattern 3...

By changing the DP confgiration as below, DP 100 has more preference than DP 11.

>dial-peer voice 11 pots
> destination-pattern 3013
> preference 1
> port 2/1

>dial-peer voice 100 voip
> destination-pattern 3013

But, incoming call from PBX will loop and will route back to PBX. To avoid,

>Config t

>dial-peer cor custom
> name a
> name b

>dial-peer cor list inPBX
> member a

>dial-peer cor list outPBX
> member b

>dial-peer voice 100 voip
> destination-pattern 3013
> corlist incoming inPBX
> corlist outgoing outPBX

If you apply the above configuration.

++ if the extnesion from port 0 dials port 1 by 3013, it will take DP 100 VOIP and send it to PBX (by default DP has highest preference, 0)
++ If the call is coming from PBX, it will take inbound DP 3013, because of 'destination-pattern 3013'. But, it will not select DP 100 as out going DP because of COR configuration and it will use DP 11.

Thanks,
Anil

[campbellj1]

That looks great! - I wont be back at the unit until tomorrow so will try that and let you all know.

I do have a little query tho.  Changing DP 100 from 3... to 3013 will then stop other 3xxx based numbers being dialed from the gateway to the PBX.  We have (at the moment) 50 IP handsets fed directly from the PBX that are all assigned 3xxx numbers.  How would Port 0 reach these?

Do I have to create another DP and put in 3... however doing this effectively undoes what you suggested?