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help with dial peer and E1 configuration

Antonio Brandao
Level 1
Level 1

Hi all,

I have a E1 working on my router.

And I would like to make work the follow conf

  • my range of pstn public number pool are 222 692 700 to 222 692 799
  • channels from 1 up to 10 are my inbound channels to incoming calls
  • the rest of avaiable channels are to outgoing calls for the follow standart patterns
  • 91*, 92*, 93*, 94*, 95*, 96*, 97*, 98* for mobile calls and 22* for fixed lines

Some body have some idea how to do that ??

regards

AB

1 Accepted Solution

Accepted Solutions

antonio.brandao wrote:

hey paolo,

I really need help help here if you don t want help shut **** up and leave people that want others.

I want the commands or some example not a stupid answer like yours!                                           

Excuse me but I don't get it.

1st, I told you all what you have to do, replace a port name with another.

But, you want examples of which you can find tens, with comment explanations.

These are the great work of my former colleagues at Documentation and TAC of Cisco.

They would help you immensely if you take the time to search and read documentation.

Just like any other network engineer in this industry. Or you have to be different ?

In the process of doing so, you use bad language directed to me, because "you want the commands".

Can only say good luck really.

View solution in original post

12 Replies 12

paolo bevilacqua
Hall of Fame
Hall of Fame

Do you know anything about dial-peer configuration ?

If you are not experienced enough, would not be better to get someone that is, for the task ?

I have some experience on dial peers but never had to do handling channels on a E1 interface , I would some help to start on it

AB

Just use the E1 voice port name and there is no other difference.

hey paolo,

I really need help help here if you don t want help shut fuck up and leave people that want others.

I want the commands or some example not a stupid answer like yours!                                           

antonio.brandao wrote:

hey paolo,

I really need help help here if you don t want help shut **** up and leave people that want others.

I want the commands or some example not a stupid answer like yours!                                           

Excuse me but I don't get it.

1st, I told you all what you have to do, replace a port name with another.

But, you want examples of which you can find tens, with comment explanations.

These are the great work of my former colleagues at Documentation and TAC of Cisco.

They would help you immensely if you take the time to search and read documentation.

Just like any other network engineer in this industry. Or you have to be different ?

In the process of doing so, you use bad language directed to me, because "you want the commands".

Can only say good luck really.

Paolo,

AGAIN if you want help me, good !!!! I apreciate it, if not, stop with this .... and forget it  !

Not first time I´m using this forum so don´t tell me how I must behave overhere.

If I came to this forum to ask something , I don´t just the commands I hope will get learn something and some useful information here not a greatest hint like yours.

For me or you don´t know it and want mess up somebody or you don´t want share your knowledgement with others that aren´t "GENIUS" that you are.

Well !!! ignoring this stupid discussion, is there somebody else that really help me ?

I found some docs on cisco.com and using google.com but not like I want to do.

I would like some example or guide to start it.

AB

Antonio,

Here is an example of an E1 gateway configured for SIP-ISDN on an E1 in the UK:

1. Set up the T1/E1 card as an E1

card type e1 0 0

2. E1 controller with 30 channels enabled. In this example the Service Provider doesn't use CRC-4 framing

controller E1 0/0/0
framing NO-CRC4
pri-group timeslots 1-31

3. Tweak ISDN timers as required.  I wouldn't change anything on the ISDN timers, unless otherwise required by your SP.

interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn timer T309 400000
isdn incoming-voice modem
isdn T309-enable
isdn send-alerting
isdn sending-complete
no cdp enable

4. Setup your locale for ring tone generation, cadence etc

voice-port 0/0/0:15
cptone GB
bearer-cap Speech

5. Link an outgoing dial-peer to the E1 you've just configured using the port command:

dial-peer voice 10 pots
description POTS talking dial peer for E1 #0

translation-profile incoming ABC
translation-profile outgoing ZYX
preference 3
destination-pattern .+
incoming called-number .+
direct-inward-dial
port 0/0/0:15

and that's it.

As far as creating the dial-peers for your dial-plan, you need to understand

a. translation profiles that use

b. translation rules that use

c. Cisco's regular expressions

d. Destination patterns within the dial-peer and configuration of the above elements

Adam

Adam,

Thanks by your reply, and sorry by my mistake I´m using a R2 signaling E1 circuit.

I would like to manage the channels some channels will be for incoming calls from my range of numbers and others will be for outgoing calls to pstn.

Is possible do it ? How to do tha on E1 R2 digital link, that I´m trying to do.

For example channels 1 - 10 will be reserved for incoming calls and others except 15 will be used for outgoing calls.

For outgoing I presume I will used dialpeers with same dest pattern with preference command pointing for the ports , I don´t know if can use a range of ports on dial peer. Still no sure if is it the correct way.

And no idea how separate ports form incoming calls.

Some idea ?

AB

Hi,

You need to reach out to somebody who's familar with this in your region, but in general you need to look at the controller configuration and create a DN group.i.e.

controller e1 0/0/0

ds0-group - timeslots 1-31 type r2-digital r2-compelled ani   etc

cas-custom 0

country ....

!

there was somebody in Brazil troubleshooting this the other day.

look at

https://supportforums.cisco.com/thread/251795?tstart=0

As far as negotiated b-channels are concerned, again please check with somebody in your region. In the UK this is acheived in configuration under the serial interface (ISDN bhan command)

Adam

Yes , not usual in UK and US. Most used on other parts of europe, australia, and south america.

I will try find somebody in these places to find a solution

Thanks for you help, I really apreciate

AB

Well,

Since I'm interested, I've had a quick play on an old router, it looks like you may have to create multiple DS groups on the controller, eg number 1 and number 2

You could then use ds group 2 for your outgoing calls, and gruop 1 for incoming.

I'm not sure how you controll which DS0 is seized from within the group though.

I have:

controller E1 3
shutdown
clock source line secondary 3
ds0-group 1 timeslots 1-10 type r2-digital
ds0-group 2 timeslots 11-15,17-31 type r2-digital
!

and then on the dial-peer:

as5300.sov1(config)# dial-peer voice 5 pots

as5300.sov1(config-dial-peer)#port 3:?

     <1-2>  DS0 Group #

so you should be able to split your outgoing calls that way.

regards

Hi All,

Again with this issue,

I made it work for outgoing calls and to I´m still getting some issue for incoming calls.

I have a extension 6510 registered at one CME inside my network that I want make receive and make calls.

Between CM and gateway I´m using H323 and both are reacheble.

Follow some parts of my conf and my out from debug voice ccapi inout command

Can somebody help with that, some little problem causing, but I can´t see what is ??!!!!

controller E1 0/0/1
framing NO-CRC4
line-termination 75-ohm
ds0-group 0 timeslots 1-10 type r2-digital r2-compelled
ds0-group 1 timeslots 11-15,17-31 type r2-digital r2-compelled ani
cas-custom 1
description LAD PSTN 1

voice-port 0/0/1:0
translation-profile incoming changeto6510

voice translation-profile changeto6510
translate called 10

voice translation-rule 10
rule 1 /222692700/ /6510/

dial-peer voice 201 pots
huntstop
destination-pattern 922T
direct-inward-dial
port 0/0/1:1
forward-digits 9
!
dial-peer voice 301 voip
destination-pattern 6T
voice-class codec 1
voice-class h323 1
session target ipv4:146.42.39.210
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
no vad
!
dial-peer voice 10 pots
incoming called-number .
direct-inward-dial
!
dial-peer voice 202 pots
huntstop
destination-pattern 99[1-3]T
direct-inward-dial
port 0/0/1:1
forward-digits 9
!

LOG - debug voip ccapi inout


*Sep 13 18:00:38.880 GMT+1: //-1/44D4E6628780/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=6510
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0
*Sep 13 18:00:38.880 GMT+1: //-1/44D4E6628780/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x662ABEA8, Call Info(
   Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=6510(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subsriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=10, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
   Sub Calling(TON=National, NPI=Telex, Data=_Hh oD, Length=101)
   Sub Called(TON=Abbreviated, NPI=Data, Data=      , Length=160)
*Sep 13 18:00:38.880 GMT+1: //-1/44D4E6628780/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Sep 13 18:00:38.880 GMT+1: //-1/44D4E6628780/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Sep 13 18:00:38.880 GMT+1: //992/44D4E6628780/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=6510(TON=Unknown, NPI=Unknown))
*Sep 13 18:00:38.880 GMT+1: //992/44D4E6628780/CCAPI/cc_process_call_setup_ind:
   Event=0x661AE138
*Sep 13 18:00:38.880 GMT+1: //992/44D4E6628780/CCAPI/ccCallSetContext:
   Context=0x6821AAD0
*Sep 13 18:00:38.880 GMT+1: //992/44D4E6628780/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 992 with tag 10 to app "_ManagedAppProcess_Default"
*Sep 13 18:00:38.880 GMT+1: //992/44D4E6628780/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
*Sep 13 18:00:38.884 GMT+1: //992/44D4E6628780/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=FALSE, Mode=0,
   Outgoing Dial-peer=301, Params=0x6821B760, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
*Sep 13 18:00:38.884 GMT+1: //992/44D4E6628780/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Sep 13 18:00:38.884 GMT+1: //992/44D4E6628780/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Sep 13 18:00:38.884 GMT+1: //992/44D4E6628780/CCAPI/ccCallSetupRequest:
   Destination Pattern=6T, Called Number=6510, Digit Strip=FALSE
*Sep 13 18:00:38.884 GMT+1: //992/44D4E6628780/CCAPI/ccCallSetupRequest:
   Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=6510(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=
   Account Number=, Final Destination Flag=TRUE,
   Guid=44D4E662-BE8F-11DF-8780-CACEBA595EE8, Outgoing Dial-peer=301
*Sep 13 18:00:38.884 GMT+1: //992/44D4E6628780/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=6510
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0
*Sep 13 18:00:38.884 GMT+1: //992/44D4E6628780/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x65A45E60, Interface Type=1, Destination=, Mode=0x0,
   Call Params(Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=6510(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subsriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=301, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
   Sub Calling(TON=National, NPI=Telex, Data=, Length=101)
   Sub Called(TON=Abbreviated, NPI=Data, Data=      , Length=160)
*Sep 13 18:00:38.884 GMT+1: //993/44D4E6628780/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=1, FlowMode=1
*Sep 13 18:00:38.884 GMT+1: //993/44D4E6628780/CCAPI/ccCallSetContext:
   Context=0x6821B710
*Sep 13 18:00:38.884 GMT+1: //992/44D4E6628780/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=301
*Sep 13 18:00:38.888 GMT+1: //993/44D4E6628780/CCAPI/cc_api_call_disconnected:
   Cause Value=38, Interface=0x66068BDC, Call Id=993
*Sep 13 18:00:38.888 GMT+1: //993/44D4E6628780/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=38, Retry Count=0)
*Sep 13 18:00:38.888 GMT+1: //993/44D4E6628780/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
*Sep 13 18:00:38.888 GMT+1: //993/44D4E6628780/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=993
*Sep 13 18:00:38.888 GMT+1: //993/44D4E6628780/CCAPI/ccCallDisconnect:
   Cause Value=38, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=38)
*Sep 13 18:00:38.888 GMT+1: //993/44D4E6628780/CCAPI/ccCallDisconnect:
   Cause Value=38, Call Entry(Responsed=TRUE, Cause Value=38)
*Sep 13 18:00:38.888 GMT+1: //993/44D4E6628780/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
*Sep 13 18:00:38.888 GMT+1: //993/44D4E6628780/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x66068BDC, Tag=0x0, Call Id=993,
   Call Entry(Disconnect Cause=38, Voice Class Cause Code=0, Retry Count=0)
*Sep 13 18:00:38.888 GMT+1: //993/44D4E6628780/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*Sep 13 18:00:38.888 GMT+1: //992/44D4E6628780/CCAPI/ccCallDisconnect:
   Cause Value=38, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Sep 13 18:00:38.888 GMT+1: //992/44D4E6628780/CCAPI/ccCallDisconnect:
   Cause Value=38, Call Entry(Responsed=TRUE, Cause Value=38)
*Sep 13 18:00:38.888 GMT+1: //992/44D4E6628780/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
*Sep 13 18:00:41.272 GMT+1: //992/44D4E6628780/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x662ABEA8, Tag=0x0, Call Id=992,
   Call Entry(Disconnect Cause=38, Voice Class Cause Code=0, Retry Count=0)
*Sep 13 18:00:41.272 GMT+1: //992/44D4E6628780/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent

Regards

AB

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