call forward noan and busy

Unanswered Question
Mar 4th, 2007

I have configured an IP phone (extension 100) for receiving all the incoming calls of my CME 3.3 Then I have configured it to forward calls to another IP phone ( extension 112 )when it's busy or no answered.

The problem is that this command only runs when you call from an other internal extension. i.e. if I call from extension 101 to 100 and it's no answered it forward to 112 correctly.

But the problem is when you call from the PSTN, you call to 964812530, this number is converted to 100, the extension 100 rings, but when it's not answered or busy doesn't jump to 112.

How can I fix it?

This is my config:

voice translation-rule 1

rule 1 /1../ /964812530/

!

voice translation-rule 3

rule 1 /964812530/ /100/

!

!

voice translation-profile SIPout

translate calling 1

!

voice translation-profile incoming

translate called 3

!

!

voip-incoming translation-profile incoming

!

!

!

!

control-plane

!

!

!

!

!

!

!

!

dial-peer voice 100 voip

translation-profile outgoing SIPout

destination-pattern 9........

session protocol sipv2

session target sip-server

codec g711ulaw

!

dial-peer voice 101 voip

translation-profile outgoing SIPout

destination-pattern 6........

session protocol sipv2

session target sip-server

codec g711ulaw

!

dial-peer voice 102 voip

translation-profile outgoing SIPout

destination-pattern 00.T

session protocol sipv2

session target sip-server

codec g711ulaw

!

sip-ua

authentication username xxx password xxx

no remote-party-id

retry invite 4

retry response 3

retry bye 2

retry cancel 2

retry register 5

timers register 250

registrar ipv4:213.x.201.x expires 60

sip-server ipv4:213.x.201.x

!

!

gatekeeper

shutdown

!

!

telephony-service

max-ephones 18

max-dn 20

ip source-address 192.168.3.1 port 2000

auto assign 1 to 18

system message Accesos Web Alternativos AWA

url services http://phone-xml.berbee.com/menu.xml

user-locale ES

network-locale ES

create cnf-files version-stamp Jan 01 2002 00:00:00

max-conferences 4 gain -6

moh en_bacd_music_on_hold.au

transfer-system full-consult

transfer-pattern .T

!

!

ephone-dn 1 dual-line

number 107 secondary 964812501 no-reg primary

name Angel Gallardo

!

!

ephone-dn 2 dual-line

number 101 secondary 964812535 no-reg primary

name Nuria Font

!

!

ephone-dn 3 dual-line

number 102 secondary 964812537 no-reg primary

name Jose Seijas

!

!

ephone-dn 4 dual-line

number 103 secondary 964812539 no-reg primary

name Oscar Pinto

!

!

ephone-dn 5 dual-line

number 104 secondary 964812536 no-reg primary

name Pepe Gallardo

!

!

ephone-dn 6 dual-line

number 105 secondary 964812538 no-reg primary

name Pablo Castell

!

!

ephone-dn 7 dual-line

number 111 no-reg primary

name diseno

!

!

ephone-dn 8 dual-line

number 100 no-reg primary

name Caiana

call-forward busy 112

call-forward noan 112 timeout 10

no huntstop

!

!

ephone-dn 9 dual-line

number 108 secondary 964812502 no-reg primary

name Miguel Campos

!

!

ephone-dn 10 dual-line

number 109 no-reg primary

!

!

ephone-dn 11 dual-line

number 110 no-reg primary

name Comerciales

!

!

ephone-dn 12 dual-line

number 106 no-reg primary

name Bea

!

!

ephone-dn 13 dual-line

number 112 no-reg primary

name Raquel

!

--More--

I have this problem too.
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rhugo Sun, 03/04/2007 - 06:18

I didn't see any of your circuit detail, but if this is a PRI with DIDs, try removing your voice translation-rule 1 and adding the "dialplna-pattern" which will act similar to the "num-exp"..... then set up a "ephone-hunt 1 sequential" with 100, followed by 101, then send the call to the next extension that can provide coverage. I think you will find that the hunt feature is more suited to what you are trying to do.

Good Luck!

sarenos2006 Sun, 03/04/2007 - 11:18

I don't have PRI, It's connected to a SIP provider.

Can you give a more detailed example of what you are saying?

I'd like to try it but I haven't understand it well.

Paolo Bevilacqua Sun, 03/04/2007 - 13:42

Hola sarenos,

the thing is that when you enable CF and the call in coming via SIP, the router will send a "302 moved temporarily" message with the forwarded extension number. Reasonably, the ITSP will not rehunt to this extension number that is unknown to his dialing plan.

Starting with CME 4.1 (IOS 12.4(11)XJ) you can disable this globally under "voice service voip" or on a per-dial-peer basis with "no supplementary-service sip moved-temporarily", and the transfer will happen transparently to the SIP calling party.

The suggestion about using hunt group I think is reasonable if you constantly use CF on these phone-dn.

Also note, you don't need translation incoming, because you have configured secondary numbers on the ephone-dn.

Remember to rate useful posts!

sarenos2006 Sun, 03/18/2007 - 04:48

Hi, I have now a new cisco router 3825 with (IOS Version 12.4(13r)T ) I think the IOS version is correct for doing what you are saying.

But the problem is it has come with CCME 3.3 so I want to upgrade to CCME 4.1

Please, How can I upgrade to CCME 4.1? Could you explain me? Is it possible to upgrade by the USB ports ?

regards.

john.caston Sun, 03/04/2007 - 22:43

Hi,

I had a similar issue when connecting a Linksys system to our CCME via a SIP trunk, even though it would ring extensions fine it wasn't being diverted to voicemail. I can't recall exactly what the problem was but I think it was to do with the fact that the phones were running Cisco's standard SCCP protocol, so I think there was something wrong with the translation between SIP and SCCP. I think I loaded SIP firmware on the phones and then it worked OK - not 100% sure?

I also have the same problem between two branches each with CCME and CUE - I think it's the same issue, even though the calls go through to the phone they don't reach the CUE (which is a SIP trunk from the CCME)?

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