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2811+CCME

nathan.murr
Level 1
Level 1

Hello,

I have 2811 router (4FXO+4FXS+2FE+built in 16-port switch)

I want to complete the following tasks:

1) Link the incoming CO-line to one of the ip-phones:

I'm not sure, but i think it's enough to do this:

voice-port 0/0/1

connection plar opx <number of ip-phone>

But in some examples i've found that i also need to do this:

dial-peer voice 110 pots

destination-pattern 82

port 0/0/1

!

!

ephone-dn 4

number 8889

trunk 82

Can anyone please explain what are the "destination pattern 82" and "trunk 82" for?

2) Apply the auto-attendant script to the CO-line, so the caller can dial additional number to reach the ip-phones.

I've setup my router according to the following guide:

http://www.cisco.com/en/US/docs/ios/voice/ivr/configuration/guide/ivrapp02.html

So i've entered the following:

application

service callme flash:simpleCall.vxml

dial-peer voice 1 pots

service callme

incoming called-number 5550121 // Access number to dial into gateway

But i did not achieve a sucsess.

When calling to the router from the CO-line, the script answers and suggests to enter the additional number (4digits). After entering the digits - i hear short tones.

The script was downloaded from the same guide.

3) Link the co-line to one of the phones, attached to one of the FXS-ports.

Is it enough:

dial-peer voice 1 pots

incoming called-number <number of CO-line>

port <number of FXS port>

Thank you.

Any advice or assistance would be really appreciated.

8 Replies 8

Marwan ALshawi
VIP Alumni
VIP Alumni

hi Oleg

about ur first question se the foolowing link which is a case i have addressed same as yours

and see all the posts on this disscustion because it is helpful for ur case and i have posted there two diffrent ways to do it:

http://forums.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=IP%20Telephony&topicID=.ee6c829&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40%5E1%40%40.2cc10da6/6#selected_message

about second one not sure!

about third question

let say ur FXO port is 0/0/2

and FXS 0/1/0

and the line number on the fxs is 123

make first pots dial peer that gives this number to the fxs

dial-peer voice 123 pots

destination-pattren 123

port 0/1/0

now go to the FXO to make forward any incoming number to the fxs

voice-port 0/0/2

connection plar opx 123

if u wanna only the fxs to use this fxo port u can achieve it by useing COR list

good luck

please, if helpful Rate

Thanks for your help.

1) Now it sees to be clear regarding the first question.

But a new problem arose:

For example, i have 4 incoming CO-lines (for each of the FXO ports). How to make a dial peer, that will choose a non-busy fxo port for a destination pattern 9T ?

2) So you don't have any ideas how to do this or you have but you're not sure? If you have any - please post, i'll just try it.

3) Thanks, now i've almost got it.

But why my example won't work?

I mean this:

dial-peer voice 1 pots

incoming called-number

port

And please, can you be more specific - what is a COR list and how to use it?

hi i am glad ur view getting better

ok i will star from question 3:

have a look at the following link it is one of my post regarding COR list which is helpful for u to understand it (cor list ) let u make level of permision and previliges based on phone or dial-peer :

http://forums.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=General&topicID=.ee6c82b&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40%5E1%40%40.2cc0badc/2#selected_message

about ur question one

fir i have a suggestion for u

with a dial-peer its destination pattren 9T this one will take long time tp be prossed as the T mean any digits and the system will wait the defual inter digit timeout which is 15 second to consider this number has finished

u better to make it more spesific like if u want the dial peer to dial local calls and the number will be 6digits make like

9......

and one for international 10 digist

9..........

ok this is for better performnace only

now if u wanna redundant dia-peer ur friend here is the prefrence command this commnad make the more prefered dial-peer with the same distination number the lower preference number the prefered

lets say u ahve three FXO ports 0/0/1 0/0/2 0/0/3 and the dial will be in the sequence first port if busy go to second then third

do the folowing :

dial-peer voice 9 pots

destination-pattren 9T

prefrence 0

port 0/0/1

dial-peer voice 92 pots

destination-pattren 9T

prefrence 2

port 0/0/2

dial-peer voice 93 pots

destination-pattren 9T

prefrence 3

port 0/0/3

about question 2 give me time to check it up becuase i have no idea about it yet

good luck

if helpful Rate

Thanks a lot, now it seems to be all clear except question 2. I'm willing to receive a reply from you.

Looks like i have to use b-acd tcl script in order to complete task number 2.

dial-peer voice 123 pots

service aa

incoming-called number .....

port 0/0/1

In the guide they say that this peer will transfer all incoming calls on port 0/0/1 to b-acd pilot number which is here "incoming-called number".

I don't get it - what pilot number?

I've set the voicexml script according to the guide:

http://www.cisco.com/en/US/docs/ios/voice/ivr/configuration/guide/ivrapp02.html

The script answers when calling to fxo 0/0/0, but when it says "Enter the destination number" - and when i enter it - i hear short tones.

The debug says:

ERROR: Destination URI is not supported.

What's "URI" and how to resolve this issue?

Hmm...now i'm getting this:

Parse Error: url_parseTelUrl: Error - Local subscriber is

missing phone-context in tel url:

1234

*Oct 1 13:27:13.058: //-1/79767AD283E4/DPM/dpMatchPeersCore:

Calling Number=, Called Number=tel: 1234, Peer Info Type=DIALPEER_INFO_SPEECH

*Oct 1 13:27:13.058: //-1/79767AD283E4/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=tel: 1234

*Oct 1 13:27:13.058: //-1/79767AD283E4/DPM/dpMatchPeersCore:

No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

there's a voice class URI tag TEL/SIP command. I think somehow it's related to my issue. Can anyone please explain how to use this voice class?

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