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Helpful
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Replies

WebEx Dial Plan to FXO

rtarson
Level 1
Level 1

Hello,

 

So I am trying to configure phone gateway we have at a location that is directly attached analog to a old paging system. Which activates on a #25. Right now in WebEx I have the gateway CUBE connected as trunk to WebEX with the DialCast 31110. I have created a voice translation rule in the gateway for 31110 to go to #25

voice translation-rule 202
rule 1 /^31110$/ /#25/
!
!
voice translation-profile TRANSLATE31110
translate called 202

 

I then created a Voip DialPeer as seen below

dial-peer voice 3 voip
description Paging-in-VOIP
translation-profile outgoing TRANSLATE31110
max-conn 250
answer-address 31110
session protocol sipv2
incoming uri request 200
voice-class codec 99
voice-class stun-usage 200
no voice-class sip localhost
dtmf-relay rtp-nte
srtp
no vad
!

Then I have a dial-peer pots that is set for destination-pattern #25

dial-peer voice 2 pots
destination-pattern #25
port 0/1/1
no sip-register
!

 

I have played around with changing called-party, answer-address ect... This configuration I atleast got better feedback with a  No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1).

I'm a little rusty when it comes to configuring dial-peer expecially going to a FXO Pots line, the tips or suggestions would be greatly appreciated. I have also provided a bigger snippet of the error I recieved when I tried to dial the dialcast incase it helps.

 

 

Jun 10 14:49:24.423 EDT: //-1/FDA1F1EAA352/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x7FB9CDFDF8C8, Call Info(
   Calling Number=182136,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=31110(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=3, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=13427
Jun 10 14:49:24.423 EDT: //-1/FDA1F1EAA352/CCAPI/ccCheckClipClir:
   In: Calling Number=182136(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Jun 10 14:49:24.423 EDT: //-1/FDA1F1EAA352/CCAPI/ccCheckClipClir:
   Out: Calling Number=182136(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Jun 10 14:49:24.423 EDT: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jun 10 14:49:24.423 EDT: :cc_get_feature_vsa malloc success
Jun 10 14:49:24.423 EDT: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jun 10 14:49:24.423 EDT:  cc_get_feature_vsa count is 1
Jun 10 14:49:24.423 EDT: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Jun 10 14:49:24.424 EDT: :FEATURE_VSA attributes are: feature_name:0,feature_time:140436022724940,feature_id:18
Jun 10 14:49:24.424 EDT: //13427/FDA1F1EAA352/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=182136(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=31110(TON=Unknown, NPI=Unknown))
Jun 10 14:49:24.424 EDT: //-1/FDA1F1EAA352/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x7FB9DC4B4BD0; count=1
Jun 10 14:49:24.424 EDT: //-1/FDA1F1EAA352/RXRULE/regxrule_stack_pop_callinfo_internal: numinfo=0x7FB9DC50A9A8
Jun 10 14:49:24.424 EDT: //13427/FDA1F1EAA352/CCAPI/cc_process_call_setup_ind:
   Event=0x7FB9CF419588
Jun 10 14:49:24.425 EDT: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 31110
Jun 10 14:49:24.425 EDT: //13427/FDA1F1EAA352/CCAPI/ccCallSetContext:
   Context=0x7FB9C761AD68
Jun 10 14:49:24.425 EDT: //13427/FDA1F1EAA352/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 13427 with tag 3 to app "_ManagedAppProcess_Default"
Jun 10 14:49:24.425 EDT: //-1/FDA1F1EAA352/RXRULE/regxrule_stack_push_RegXruleNumInfo_internal: stack=0x7FB9DC4B4BD0; count=1
Jun 10 14:49:24.425 EDT: //13427/FDA1F1EAA352/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
Jun 10 14:49:24.425 EDT: //-1/FDA1F1EAA352/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=31110, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 10 14:49:24.425 EDT: //-1/FDA1F1EAA352/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=31110
Jun 10 14:49:24.425 EDT: //-1/FDA1F1EAA352/DPM/dpMatchPeersCore:

Jun 10 14:49:24.426 EDT: Match Rule=DP_MATCH_DEST_SIP_USER; Called Number=31110
Jun 10 14:49:24.426 EDT: //-1/FDA1F1EAA352/DPM/dpMatchPeersCore:
   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
Jun 10 14:49:24.426 EDT: //-1/FDA1F1EAA352/DPM/dpMatchSafModulePlugin:
   dialstring=31110, saf_enabled=0, saf_dndb_lookup=1, dp_result=-1
Jun 10 14:49:24.426 EDT: //-1/FDA1F1EAA352/DPM/dpMatchPeersMoreArg:
   Result=NO_MATCH(-1)
Jun 10 14:49:24.426 EDT: //-1/FDA1F1EAA352/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=31110, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 10 14:49:24.426 EDT: //-1/FDA1F1EAA352/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=31110
Jun 10 14:49:24.426 EDT: //-1/FDA1F1EAA352/DPM/dpMatchPeersCore:

Jun 10 14:49:24.426 EDT: Match Rule=DP_MATCH_DEST_SIP_USER; Called Number=31110
Jun 10 14:49:24.426 EDT: //-1/FDA1F1EAA352/DPM/dpMatchPeersCore:
   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
Jun 10 14:49:24.426 EDT: //-1/FDA1F1EAA352/DPM/dpMatchSafModulePlugin:
   dialstring=31110, saf_enabled=0, saf_dndb_lookup=1, dp_result=-1
Jun 10 14:49:24.426 EDT: //-1/FDA1F1EAA352/DPM/dpMatchPeersMoreArg:
1 Accepted Solution

Accepted Solutions

The old paging system I was working with is an old Dukane system. That has a analog line via FXO to the Dukane system that triggers a relay for live audio via specific dial tone which was #25 for this scenario. 

 

I was able to get it working. Here is the config I used to get the paging to work over webex and to translate the incoming dial into the code for live audio, in case others stumble upon the same quest here you are:

Voice Service Voip:

voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
media statistics
media bulk-stats
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip refer
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711ulaw
trace
stun
stun flowdata agent-id 1 boot-count 4
stun flowdata shared-secret ****
sip
asymmetric payload full
early-offer forced
g729 annexb-all

Translation Rules:
voice translation-rule 202
rule 1 /^31110$/ /#25/
!
!
voice translation-profile TRANSLATE31110
translate called 202

voice class codec 99
codec preference 1 g711ulaw
codec preference 2 g711alaw

Voice Port and Dial Peers

voice-port 0/1/1
caller-id enable

dial-peer voice 2 pots
translation-profile outgoing TRANSLATE31110
destination-pattern 31110
port 0/1/1
no sip-register


dial-peer voice 3 voip
description VOIP-Paging
translation-profile outgoing TRANSLATE31110
max-conn 250
answer-address 31110
session protocol sipv2
incoming called-number 31110
incoming uri request 200
voice-class codec 99
voice-class stun-usage 200
no voice-class sip localhost
dtmf-relay rtp-nte
srtp
no vad

 

 

 

 

 

View solution in original post

3 Replies 3

eyalraba
Cisco Employee
Cisco Employee

@rtarson ,

It should be translation-profile incoming TRANSLATE31110 instead of outgoing. Since you are receiving the call from Webex Calling, i.e. inbound, the profile with you configured will not take affect.

SteveK066
Level 1
Level 1

Just to verify - the paging system is providing dial tone, correct? Valcom and Viking page adapters can be configured either way. With an FXO, it would connect to a trunk port on a legacy system. The trunk access code would be entered, and upon hearing a dial tone, a 'zone' would be selected, or typically '0' for all zones. Unlike an FXS port, which would connect to an ATA, and 'answered' by the page interface when that extension is called. 

A basic interface I've done over the years is setting a VoIP phone for auto-answer, and connecting the speaker output to the input of the paging amplifier.

The old paging system I was working with is an old Dukane system. That has a analog line via FXO to the Dukane system that triggers a relay for live audio via specific dial tone which was #25 for this scenario. 

 

I was able to get it working. Here is the config I used to get the paging to work over webex and to translate the incoming dial into the code for live audio, in case others stumble upon the same quest here you are:

Voice Service Voip:

voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
media statistics
media bulk-stats
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip refer
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711ulaw
trace
stun
stun flowdata agent-id 1 boot-count 4
stun flowdata shared-secret ****
sip
asymmetric payload full
early-offer forced
g729 annexb-all

Translation Rules:
voice translation-rule 202
rule 1 /^31110$/ /#25/
!
!
voice translation-profile TRANSLATE31110
translate called 202

voice class codec 99
codec preference 1 g711ulaw
codec preference 2 g711alaw

Voice Port and Dial Peers

voice-port 0/1/1
caller-id enable

dial-peer voice 2 pots
translation-profile outgoing TRANSLATE31110
destination-pattern 31110
port 0/1/1
no sip-register


dial-peer voice 3 voip
description VOIP-Paging
translation-profile outgoing TRANSLATE31110
max-conn 250
answer-address 31110
session protocol sipv2
incoming called-number 31110
incoming uri request 200
voice-class codec 99
voice-class stun-usage 200
no voice-class sip localhost
dtmf-relay rtp-nte
srtp
no vad