12-03-2009 04:05 AM - edited 03-15-2019 08:39 PM
Hello,
I need some help regarding the topic mentioned. Let me give a brief overview of my issue.
My company has an IP Telephony setup in our office complete with a 3640 router and CUCM 6.0.1 with the router installed with two NM-2V and each module with two VIC-2FXO totaling upto 8 FXO ports.
Now my boss wants me to map a certain FXO trunk line to his extension so any incoming/outgoing call would be to/from his IP Phone. Now I can do something about the incoming calls, redirecting them to his extension using the 'connection plar' but when it comes to outgoing calls, I cannot do this (or don't know) unless I give that voice port a different destination pattern and creating new route pattern for it. If there is any way that this can be achieved with my current H.323 setup or I have to migrate to something else like MGCP, I would highly appreciate help on this.
Summarizing it all, he want another DN on his 7940 IP Phone which when picked up directly gives him the external FXO PSTN dial tone and when that PSTN number called will land the call on his DN. So far I have achieved incoming calls with connection plar on port 3/1/1.
Below is my router configuration.
interface FastEthernet1/0
ip address 192.168.10.52 255.255.255.0
duplex auto
speed auto
!
voice-port 2/0/0
timeouts interdigit 5
timeouts call-disconnect 1
timeouts wait-release 3
connection plar opx 27
caller-id enable
!
voice-port 2/0/1
timeouts interdigit 5
timeouts call-disconnect 1
timeouts wait-release 3
connection plar opx 27
caller-id enable
!
voice-port 2/1/0
timeouts interdigit 5
timeouts call-disconnect 1
timeouts wait-release 3
connection plar opx 27
caller-id enable
!
voice-port 2/1/1
timeouts interdigit 5
timeouts call-disconnect 1
timeouts wait-release 3
connection plar opx 27
caller-id enable
!
voice-port 3/0/0
timeouts interdigit 5
timeouts call-disconnect 1
timeouts wait-release 3
connection plar opx 27
caller-id enable
!
voice-port 3/0/1
timeouts interdigit 5
timeouts call-disconnect 1
timeouts wait-release 3
connection plar opx 27
caller-id enable
!
voice-port 3/1/0
timeouts interdigit 5
timeouts call-disconnect 1
timeouts wait-release 3
connection plar opx 27
caller-id enable
!
voice-port 3/1/1
timeouts interdigit 5
timeouts call-disconnect 1
timeouts wait-release 3
connection plar opx 300
caller-id enable
!
!
dial-peer voice 1 voip
destination-pattern ..
session target ipv4:192.168.10.50
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 2 pots
service script1
destination-pattern 9T
port 2/0/0
!
dial-peer voice 3 pots
service script1
destination-pattern 9T
port 2/0/1
!
dial-peer voice 4 pots
service script1
destination-pattern 9T
port 2/1/0
!
dial-peer voice 5 pots
service script1
destination-pattern 9T
port 2/1/1
!
dial-peer voice 6 pots
service script1
destination-pattern 9T
port 3/0/0
!
dial-peer voice 7 pots
service script1
destination-pattern 9T
port 3/0/1
!
dial-peer voice 8 pots
service script1
destination-pattern 9T
port 3/1/0
!
dial-peer voice 9 pots
destination-pattern 9T
port 3/1/1
Thanks in advance.
Suhaib.
Solved! Go to Solution.
12-08-2009 05:26 AM
Suhaib,
I tried this on my lab system on an ISDN port and it worked (don't have a live CO port),
As Justforvoice correctly stated you can't have an empty route pattern so:
Create a translation pattern that is empty (that IS permitted), and use the "Called Party Transform Mask" to change the number to 888
Create an 888 route pattern and point it at the gateway - pass all three digits out
On the gateway create a dial-peer with a destination-pattern of 888 - you don't need the digit-strip command since it is on by default.
The gateway should sieze the FXO and you should hear dial-tone form the CO.
HTH,
Art
12-11-2009 07:23 AM
Suhaib,
Glad to hear things are working, and hopefully your boss is happy now. Happy to help.
Art
PS, I kinda figured you would figure out the PT and CSS - good job!
12-03-2009 05:36 AM
Suhaib,
If he will agree use 8 instead of 9 as the trunk access code (on the new line only) and change the dial-peer for one of the ports to 8T. Set up a new partition and CSS for his phone that allows it and put the RP in for his use. Put in a block pattern for leading 8s in the partition for the other phones so only he can use it, and nobody else can.
HTH,
Art
12-03-2009 10:40 PM
Hello Art,
I have already tried it by defining a trunk tag on the port using 'destination-pattern 8' and making a RP to use it and was already on the way to defining the partition and CSS but told him just incase that this was what I was doing. Just by dialing 8 he will get the external dialtone. He halfheartedly agreed to use it for the 'time being' but wants it done the way I mentioned in my initial post, 'pick the extension and get the external dialtone'.
Is there any way to define the DN on the IP Phone to dial the trunk tag everytime the extension is picked up. Like when presses the extension button 8 is dialed automatically so he hears the external dialtone without the need to press 8?
Hoping for an early reply.
Suhaib.
12-04-2009 05:26 AM
Suhaib,
First my apologies for not thoroughly reading you post - this final step should be easy, and give him what he wants.
Since the second line on his phone already has it's own CSS and partition what you eant to do is alter the route pattern with the 8. Delete the 8 so that the RP is empty. When he picks up his phone it will immediately match the empty RP and should go to the port. This is how you set up a PLAR, but with a PLAR you then put the number you want the line to dial in the mask for the called number.
HTH,
Art
12-07-2009 03:05 AM
Hello again Art,
I had made the CSS and Partition on the extension and doing the needful but the RP is not accepting a blank entry. So I cannot get it done. Also the PLAR thing you mentioned. I kinda got lost on that part.
Currently what I am doing is the assigning a prefix of 8 so when any number is dialed it is added and the called is then routed to the gateway without him to dial any extra prefix digit. But that still doesn't solve the problem when he picks up that extension and gets an external PSTN dialtone.
Suhaib.
12-07-2009 04:04 AM
Blank Rp is not allowed it is not TP.
you need to do the following:
1- leave his line in different partition and CSS.
2- create new RPs for all cases like:
2XXXXXXX ( for local calls)
08XXXXXXXX ( for Mobile)
3- Do not use any access code
4- Remove provide outside dial tone from the RP.
5- On the Gateway create new dial-peers for all new RPs and map them to the FXO Line.
12-08-2009 04:28 AM
This still leaves out one problem of getting the external tone on picking up the set as when implemented, the IP Phone is just giving the regular tone.
12-08-2009 05:26 AM
Suhaib,
I tried this on my lab system on an ISDN port and it worked (don't have a live CO port),
As Justforvoice correctly stated you can't have an empty route pattern so:
Create a translation pattern that is empty (that IS permitted), and use the "Called Party Transform Mask" to change the number to 888
Create an 888 route pattern and point it at the gateway - pass all three digits out
On the gateway create a dial-peer with a destination-pattern of 888 - you don't need the digit-strip command since it is on by default.
The gateway should sieze the FXO and you should hear dial-tone form the CO.
HTH,
Art
12-11-2009 04:05 AM
Hello Art and JustForVoice,
Thanks for all your help in this regard as this has finally solved my problem with the direct line. Once again this forums has proved to have wonderfull tech-savvy people who are really helpfull.
PS: Art, you forgot to mention that inorder to create the blank translation pattern, it need to belong to a Partition. Also for that to work, it has to be associated to the respective CSS.
Regards,
Suhaib.
12-11-2009 07:23 AM
Suhaib,
Glad to hear things are working, and hopefully your boss is happy now. Happy to help.
Art
PS, I kinda figured you would figure out the PT and CSS - good job!
12-11-2009 07:27 AM
There is not need to use CSS and PT fopr this. It can be done configuring the GW only. See:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801bc341.shtml
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12-12-2009 01:05 AM
Thanks for looking in to this post p.bevilacqua. What you pointed out maps an FXO port to a specific DN but it doesn't give me direct access where I just dial the external number. The link you mentioned assumed that I am pressing 9 to dial out to the PSTN.
12-12-2009 01:45 AM
Not really, it works also without an external access digit, as convenient in many countries.
It is all based on the calling number.
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12-12-2009 06:12 AM
Suhaib,
Paolo is correct - this will work without a trunk access code of 9 or anything else since it is based on the calling line ID, but there still would have been an issue. One of your stated requirements was that your boss get outside dial tone immediately when he lifted the receiver, which would not happen here until the first digit input (if you had "provide outside dialtone" checked on all of your outside RPs). He also required that the single line work in this manner, like a key system, but your other lines worked with the normal PBX like behavior. That also pretty much puts CSS and PTs into the picture to separate the two dial plan requirements.
Nice little exercise though - keeps the brain cells stirred up ; ^))
Cheers!
Art
09-10-2012 10:45 PM
my devices is AS5350 as E1 gateway & 2811 as PSTN gateway and i have CUCM 6.1 on HP server
i want map a DN to a FXO , but i can't
2811 registered in cucm as mgcp gateway and worked correctly with several Route Pattern
but i want 1 Route Pattern for all calls
i have 7962 IP Phone , could i set a botton for a string digits?
for example , set 82 on one botton and when everyone press that botton , 82 pass to cucm
and finally
how i could hear the external FXO PSTN dial tone on a IP Phone and when PSTN number called will land the call on that?
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