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30
Helpful
13
Replies

Help - Hide Prefix Translated on Display Phone

crusier2015
Beginner
Beginner

Hi ,

I´ve implemented with success the follow translation and outbound calls are working , but  ISP have as mandatory that always i sent prefix(115555) for all outbound calls,  this prefix appears on displau phone of users. For example if user call to a local number :  0+44334433, and call works and is transformed as 11555544334433 and appears on his display phone.

Is there anyway to hide this 115555 for users, and show only destination number called?

Tks

voice translation-rule 1
rule 1 /^0/ /115555/

voice translation-profile LOCAL-PREFIX
translate called 1


dial-peer voice 20 voip
description Ligacao Local
translation-profile outgoing LOCAL-PREFIX
destination-pattern 0[1-9]T
session protocol sipv2
session target ipv4:3.3.3.3
voice-class sip dtmf-relay force rtp-nte
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
clid network-number 115555
!


voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 600 min 60

 

voice register global
mode cme
source-address 192.168.3.3 port 5060
timeouts interdigit 2
max-dn 200
max-pool 42
timezone 17
time-format 24
tftp-path flash:
create profile sync 0033183843410324
ntp-server 200.160.7.186 mode directedbroadcast
auto-register

voice register dn 1
number 201
name User
no-reg

voice register pool 1
busy-trigger-per-button 1
id mac 1111.1111.1111
type 7861
number 1 dn 1
dtmf-relay sip-notify
codec g711alaw

telephony-service
max-ephones 42
max-dn 200
ip source-address 192.168.3.3 port 2000
timeouts interdigit 3
time-zone 17
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Nov 07 2022 23:36:04

 

1 Accepted Solution

Accepted Solutions

On the changed translation not working. In your first post you mentioned that the translation was to replace the leading 0 (zero) with 115555, but now in you last response and in you debugs it shows that you do this translation, replace leading 0 (zero) with 11555511. With this i mind you would need to do this to make your rule a little better formatted.

voice translation-rule 1
 rule 1 /^0\(.*\)/ /11555511\1/

As you also have other rules I would suggest that you do  similar clean-up for these.

Apart from this you have an overlap in your configuration that negates the first occurrence of it.

*First one*
voice translation-profile CALL-IN
translate called 1

*Second one*
voice translation-profile CALL-IN
translate called 2

Also in general you can simplify you configuration by doing this.

voice translation-rule 1 ! Combine all destination patterns into one rule and put them in the proper order to translate correctly
 rule 1 /^00\([38]00\)/ /115555\1/
 rule 2 /^0\(00.*\)/ /115555\1/
 rule 3 /^00\(.*\)/ /115555\1/
 rule 4 /^0\(.*\)/ /11555511\1/
!
voice translation-profile PSTN-OUT ! New translation profile to be use for all egress calls to PSTN
 translate called 1
!
voice class e164-pattern-map 2000 ! Combine all destination patterns into one map and use this on a single outbound dial peer
 description E164 Pattern Map for called number to PSTN
  e164 00[38]00T
  e164 000[1-9]T
  e164 00[1-9]T
  e164 0[1-9]T
!
dial-peer voice 1 voip
 no session target ipv4:3.3.3.3 ! Not needed as it is used for outbound dial peers
!
dial-peer voice 20 voip
 description Ligacao call to PSTN
 translation-profile outgoing PSTN-OUT ! Use new translation profile
 no destination-pattern 0[1-9]T ! Not needed with the use of a map
 destination e164-pattern-map 2000 ! Use the combined map for all egress calls
 voice-class sip bind control source-interface GigabitEthernet0/1 ! Add missing bind statement
 no clid network-number 115555 ! Should not be needed and if it is needed to send calling number as 115555 all the time it can be added back or replaced by a translation rule that translate the calling number
!
no dial-peer voice 21 voip ! Not needed with all calls egressing out via dial peer 20
!
no dial-peer voice 22 voip ! Not needed with all calls egressing out via dial peer 20
!
no dial-peer voice 23 voip ! Not needed with all calls egressing out via dial peer 20
!
no dial-peer voice 24 voip ! Not needed with all calls egressing out via dial peer 20


voice service voip
 sip
  bind control source-interface GigabitEthernet0/0 ! Add bind staement that are needed for CME SIP phones
  bind media source-interface GigabitEthernet0/0 ! Add bind staement that are needed for CME SIP phones

I'll get back to you on the debugs once I've had time to look at them in detail.



Response Signature


View solution in original post

13 Replies 13

Ben Petroff
Beginner
Beginner

To my knowledge, there's no way to have the phone display a number other than the called number... The best way would be to push back on your PSTN provider for their bizarre way of operating.

I have never seen this as a requirement from a PSTN provider and I probably wouldn't accept this unless I had absolutely no choice.

In my experience this is a quite common thing with many service providers in many countries to use prefixes as part of the called number sent.



Response Signature


Roger Kallberg
VIP Expert VIP Expert
VIP Expert

Try with this in your configuration.

 

voice service voip
  sip
   no update caller-id

sip-ua
 no remote-party-id

 

 

If the above two options fail, and it may depending on your call flow topology please apply the following sip profile.

Remove UPDATE support from both the request and response events.

 

voice class sip-profiles 1
  request ANY sip-header Allow-Header modify "UPDATE, " ""
  response ANY sip-header Allow-Header modify "UPDATE, " ""

voice service voip
  sip
    sip-profile 1

 

 

You can get more details on this Ask the expert link as there was a discussion on a similar topic. Ask the Experts: Configuring and Troubleshooting Cisco SIP CUBE/Gateways and MGCP Gateways  



Response Signature


Roger - I believe the OP is saying the calling device displays the translated called number on the screen instead of the original dialed number. 

In CUCM, this can be fixed by set the "Always Display Original Dialed Number" to "True" under the service parameters. Do you know what is the equivalent of that in CME? 

That's how I read the OP as well and I think that there is no equivalent general setting in CME. But the suggested configuration should in essence get the same end result more or less, anyway for calls directed to PSTN.



Response Signature


crusier2015
Beginner
Beginner

Hi Roger, tks for both sugestions, unfortunately didn't worked, still appearing number on phone display.

a correction, if i apply only the follow, now on display only appears number 0:

sip-ua
no remote-party-id

 

That is a truly odd one. Maybe you could share the output from debug ccsip message and debug voip ccapi inout so that we can see what is actually sent and received?

I have a suggestion for your translation rule as the current one is kind of not greatly written. However it is not really related to your original question. Rewrite the rule to this.

voice translation-rule 1
 rule 1 /^0\(.*\)/ /115555\1/

I also wanted to ask you about this command under your outbound dial peer “clid network-number 115555”. Why do you have that? AFAIK that would send the calling number as 115555 to your service provider, is that the intent?



Response Signature


crusier2015
Beginner
Beginner

Hi Roger,
Tks for you support,  follow debug voip ccapi inout attached( sorry debug ccsip was not possible to attached, follow bellow the debug ccsip).

I tried the follow translation rule, but the outbound return busy signal:
voice translation-rule 1

 rule 1 /^0\(.*\)/ /115555\1/

About clid network-number 115555, withou this command the outbound works too, in fact i dont know the correct reason that i was using, i follow the example that i see on internet tutorials.

Follow all show run output:

voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 600 min 60
!
!
voice class sip-profiles 1
request ANY sip-header Allow-Header modify "UPDATE, " ""
response ANY sip-header Allow-Header modify "UPDATE, " ""
!
!
!
voice register global
mode cme
source-address 192.168.22.10 port 5060
timeouts interdigit 2
max-dn 200
max-pool 42
load 7861 sip78xx.14-1-1-0211-134
authenticate register
authenticate realm all
olsontimezone America/Maceio version 2.3.2-b02
timezone 17
time-format 24
tftp-path flash:
create profile sync 0080323448004054
ntp-server 1.1.1.1 mode directedbroadcast
auto-register
!

!
voice register dn 41
number 241
name Teste
no-reg

voice register pool 41
busy-trigger-per-button 1
id mac 0000.0000.0001
number 1 dn 41
dtmf-relay sip-notify
username 241 password 241
codec g711alaw

voice translation-rule 1
rule 1 /^0/ /11555511/
!
voice translation-rule 2
rule 1 /55114444440/ /201/
!
voice translation-rule 3
rule 1 /^00/ /115555/
!
voice translation-rule 4
rule 1 /^000/ /11555500/
!
voice translation-rule 5
rule 1 /^00800/ /115555800/
!
voice translation-rule 6
rule 1 /^00300/ /115555300/
!
!
voice translation-profile CALL-IN
translate called 1
!
voice translation-profile CALL-0300
translate called 6
!
voice translation-profile CALL-0800
translate called 5
!
voice translation-profile CALL-DDD
translate called 3
!
voice translation-profile CALL-DDI
translate called 4
!
voice translation-profile CALL-IN
translate called 2
!
voice translation-profile CALL-LOCAL
translate called 1


!
!
dial-peer voice 1 voip
translation-profile incoming CALL-IN
session protocol sipv2
session target ipv4:3.3.3.3
incoming called-number 55114444440
voice-class sip rel1xx disable
voice-class sip dtmf-relay force rtp-nte
no voice-class sip pass-thru headers
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
!
dial-peer voice 20 voip
description Ligacao Local
translation-profile outgoing CALL-LOCAL
destination-pattern 0[1-9]T
session protocol sipv2
session target ipv4:3.3.3.3
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
clid network-number 115555
!
dial-peer voice 21 voip
description Ligacao DDD
translation-profile outgoing CALL-DDD
destination-pattern 00[1-9]T
session protocol sipv2
session target ipv4:3.3.3.3
voice-class sip dtmf-relay force rtp-nte
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
clid network-number 115555
!
dial-peer voice 22 voip
description Ligacao DDI
translation-profile outgoing CALL-DDI
destination-pattern 000[1-9]T
session protocol sipv2
session target ipv4:3.3.3.3
voice-class sip dtmf-relay force rtp-nte
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
clid network-number 115555
!
dial-peer voice 23 voip
description Ligacao 0800
translation-profile outgoing CALL-0800
destination-pattern 00800T
session protocol sipv2
session target ipv4:3.3.3.3
voice-class sip dtmf-relay force rtp-nte
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
clid network-number 115555
!
dial-peer voice 24 voip
description Ligacao 0300
translation-profile outgoing CALL-0300
destination-pattern 00300T
session protocol sipv2
session target ipv4:3.3.3.3
voice-class sip dtmf-relay force rtp-nte
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
clid network-number 115555
!
!
!
sip-ua
!
!
!
gatekeeper
no shutdown
!
!
telephony-service
max-ephones 42
max-dn 200
ip source-address 192.168.22.10 port 2000
calling-number local
timeouts interdigit 3
time-zone 17
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Nov 09 2022 17:23:58
!
!

 

 

 

 

 

VoIP-Secoli#debug ccsip messages
SIP Call messages tracing is enabled
VoIP-Secoli#
Nov 9 22:33:41.733: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:


VoIP-Secoli#
Nov 9 22:33:43.969: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:044334433@all SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:58247;rport;branch=z9hG4bKPj7e171d13ccb54ed197e987e416c798fa
Max-Forwards: 70
From: "teste" <sip:241@all>;tag=5faadc91be4244cabf09cb2805e1d59f
To: <sip:044334433@all>
Contact: "teste" <sip:241@192.168.2.6:58247;ob>
Call-ID: 46c9f04967ee46fd9f9a8b432302a9c8
CSeq: 19403 INVITE
Route: <sip:192.168.22.10:5060;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.21.2
Content-Type: application/sdp
Content-Length: 337

v=0
o=- 3877011223 3877011223 IN IP4 192.168.2.6
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 101
c=IN IP4 192.168.2.6
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.6
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:2093286722 cname:36460839541b59f3

Nov 9 22:33:43.973: //8426/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.6:58247;rport;branch=z9hG4bKPj7e171d13ccb54ed197e987e416c798fa
From: "teste" <sip:241@all>;tag=5faadc91be4244cabf09cb2805e1d59f
To: <sip:044334433@all>
Date: Wed, 09 Nov 2022 22:33:43 GMT
Call-ID: 46c9f04967ee46fd9f9a8b432302a9c8
CSeq: 19403 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.6.3.M9
Session-ID: 00000000000000000000000000000000;remote=7ec3812c5ead52f5adabc7d00cde9275
Content-Length: 0


Nov 9 22:33:43.981: //8427/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:1155551144334433@3.3.3.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK953C7D
Remote-Party-ID: "teste" <sip:115555@192.168.22.10>;party=calling;screen=yes;privacy=off
From: "teste" <sip:115555@192.168.22.10>;tag=50B32D4-51A
To: <sip:1155551144334433@3.3.3.3>
Date: Wed, 09 Nov 2022 22:33:43 GMT
Call-ID: 66FB517D-5FB511ED-AB5FFC01-A571E559@192.168.22.10
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1727626348-1605702125-2874801153-2775704921
User-Agent: Cisco-SIPGateway/IOS-15.6.3.M9
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1668033223
Contact: <sip:115555@192.168.22.10:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-ID: 7ec3812c5ead52f5adabc7d00cde9275;remote=00000000000000000000000000000000
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 274

v=0
o=CiscoSystemsSIP-GW-UserAgent 8417 8292 IN IP4 192.168.22.10
s=SIP Call
c=IN IP4 192.168.22.10
t=0 0
m=audio 18264 RTP/AVP 8 101 19
c=IN IP4 192.168.22.10
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20

Nov 9 22:33:43.989: //8427/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK953C7D;received=200.170.192.130;rport=57236
From: "teste" <sip:115555@192.168.22.10>;tag=50B32D4-51A
To: <sip:1155551144334433@3.3.3.3>
Call-ID: 66FB517D-5FB511ED-AB5FFC01-A571E559@192.168.22.10
CSeq: 101 INVITE
Server: BR-SAO-CO1-SVV-SBC-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:1155551144334433@3.3.3.3:5060>
Content-Length: 0


Nov 9 22:33:43.989: //8427/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK953C7D;received=200.170.192.130;rport=57236
From: "teste" <sip:115555@192.168.22.10>;tag=50B32D4-51A
To: <sip:1155551144334433@3.3.3.3>;tag=as6f66231f
Call-ID: 66FB517D-5FB511ED-AB5FFC01-A571E559@192.168.22.10
CSeq: 101 INVITE
Server: BR-SAO-CO1-SVV-SBC-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:1155551144334433@3.3.3.3:5060>
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 547139848 547139848 IN IP4 3.3.3.3
s=Asterisk PBX 13.16.0
c=IN IP4 3.3.3.3
t=0 0
m=audio 19324 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Nov 9 22:33:43.993: //8426/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.6:58247;rport;branch=z9hG4bKPj7e171d13ccb54ed197e987e416c798fa
From: "teste" <sip:241@all>;tag=5faadc91be4244cabf09cb2805e1d59f
To: <sip:044334433@all>;tag=50B32E0-3CE
Date: Wed, 09 Nov 2022 22:33:43 GMT
Call-ID: 46c9f04967ee46fd9f9a8b432302a9c8
CSeq: 19403 INVITE
Require: 100rel
RSeq: 4180
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:1155551144334433@192.168.22.10>;party=called;screen=yes;privacy=off
Contact: <sip:044334433@192.168.22.10:5060>
Server: Cisco-SIPGateway/IOS-15.6.3.M9
Session-ID: ef729addfbd05acf8c190cdb88602e99;remote=7ec3812c5ead52f5adabc7d00cde9275
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 7527 1199 IN IP4 192.168.22.10
s=SIP Call
c=IN IP4 192.168.22.10
t=0 0
m=audio 18262 RTP/AVP 8
c=IN IP4 192.168.22.10
a=rtpmap:8 PCMA/8000
a=ptime:20

Nov 9 22:33:44.009: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
PRACK sip:044334433@192.168.22.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:58247;rport;branch=z9hG4bKPj5038b81d02894da980cf79d997d36e99
Max-Forwards: 70
From: "teste" <sip:241@all>;tag=5faadc91be4244cabf09cb2805e1d59f
To: <sip:044334433@all>;tag=50B32E0-3CE
Call-ID: 46c9f04967ee46fd9f9a8b432302a9c8
CSeq: 19404 PRACK
RAck: 4180 19403 INVITE
Content-Length: 0


Nov 9 22:33:44.009: //8426/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Sent:

VoIP-Secoli#SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.6:58247;rport;branch=z9hG4bKPj5038b81d02894da980cf79d997d36e99
From: "teste" <sip:241@all>;tag=5faadc91be4244cabf09cb2805e1d59f
To: <sip:044334433@all>;tag=50B32E0-3CE
Date: Wed, 09 Nov 2022 22:33:44 GMT
Call-ID: 46c9f04967ee46fd9f9a8b432302a9c8
Server: Cisco-SIPGateway/IOS-15.6.3.M9
CSeq: 19404 PRACK
Session-ID: ef729addfbd05acf8c190cdb88602e99;remote=7ec3812c5ead52f5adabc7d00cde9275
Content-Length: 0


VoIP-Secoli#
Nov 9 22:33:52.060: //8427/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK953C7D;received=200.170.192.130;rport=57236
From: "teste" <sip:115555@192.168.22.10>;tag=50B32D4-51A
To: <sip:1155551144334433@3.3.3.3>;tag=as6f66231f
Call-ID: 66FB517D-5FB511ED-AB5FFC01-A571E559@192.168.22.10
CSeq: 101 INVITE
Server: BR-SAO-CO1-SVV-SBC-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:1155551144334433@3.3.3.3:5060>
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 547139848 547139848 IN IP4 3.3.3.3
s=Asterisk PBX 13.16.0
c=IN IP4 3.3.3.3
t=0 0
m=audio 19324 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Nov 9 22:33:52.068: //8427/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:1155551144334433@3.3.3.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK9541CF0
From: "teste" <sip:115555@192.168.22.10>;tag=50B32D4-51A
To: <sip:1155551144334433@3.3.3.3>;tag=as6f66231f
Date: Wed, 09 Nov 2022 22:33:43 GMT
Call-ID: 66FB517D-5FB511ED-AB5FFC01-A571E559@192.168.22.10
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Session-ID: 7ec3812c5ead52f5adabc7d00cde9275;remote=ef729addfbd05acf8c190cdb88602e99
Content-Length: 0


Nov 9 22:33:52.068: //8426/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.6:58247;rport;branch=z9hG4bKPj7e171d13ccb54ed197e987e416c798fa
From: "teste" <sip:241@all>;tag=5faadc91be4244cabf09cb2805e1d59f
To: <sip:044334433@all>;tag=50B32E0-3CE
Date: Wed, 09 Nov 2022 22:33:44 GMT
Call-ID: 46c9f04967ee46fd9f9a8b432302a9c8
CSeq: 19403 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:1155551144334433@192.168.22.10>;party=called;screen=no;privacy=off
Contact: <sip:044334433@192.168.22.10:5060>
Supported: replaces
Server: Cisco-SIPGateway/IOS-15.6.3.M9
Session-ID: ef729addfbd05acf8c190cdb88602e99;remote=7ec3812c5ead52f5adabc7d00cde9275
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 7527 1199 IN IP4 192.168.22.10
s=SIP Call
c=IN IP4 192.168.22.10
t=0 0
m=audio 18262 RTP/AVP 8
c=IN IP4 192.168.22.10
a=rtpmap:8 PCMA/8000
a=ptime:20

Nov 9 22:33:52.076: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

VoIP-Secoli#ACK sip:044334433@192.168.22.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:58247;rport;branch=z9hG4bKPjc06d9e0b606642a492a7132a3ee6ffa1
Max-Forwards: 70
From: "teste" <sip:241@all>;tag=5faadc91be4244cabf09cb2805e1d59f
To: <sip:044334433@all>;tag=50B32E0-3CE
Call-ID: 46c9f04967ee46fd9f9a8b432302a9c8
CSeq: 19403 ACK
Content-Length: 0


VoIP-Secoli#
Nov 9 22:33:55.738: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:


Nov 9 22:33:56.266: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:044334433@192.168.22.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:58247;rport;branch=z9hG4bKPj38035e133c584113945cb48c1a6059fd
Max-Forwards: 70
From: "teste" <sip:241@all>;tag=5faadc91be4244cabf09cb2805e1d59f
To: <sip:044334433@all>;tag=50B32E0-3CE
Call-ID: 46c9f04967ee46fd9f9a8b432302a9c8
CSeq: 19405 BYE
User-Agent: MicroSIP/3.21.2
Content-Length: 0


Nov 9 22:33:56.270: //8426/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.6:58247;rport;branch=z9hG4bKPj38035e133c584113945cb48c1a6059fd
From: "teste" <sip:241@all>;tag=5faadc91be4244cabf09cb2805e1d59f
To: <sip:044334433@all>;tag=50B32E0-3CE
Date: Wed, 09 Nov 2022 22:33:56 GMT
Call-ID: 46c9f04967ee46fd9f9a8b432302a9c8
Server: Cisco-SIPGateway/IOS-15.6.3.M9
CSeq: 19405 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=441,OS=70560,PR=611,OR=97760,PL=0,JI=0,LA=0,DU=4
Session-ID: ef729addfbd05acf8c190cdb88602e99;remote=7ec3812c5ead52f5adabc7d00cde9275
Content-Length: 0


Nov 9 22:33:56.270: //8427/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:1155551144334433@3.3.3.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK9551180
From: "teste" <sip:115555@192.168.22.10>;tag=50B32D4-51A
To: <sip:1155551144334433@3.3.3.3>;tag=as6f66231f
Date: Wed, 09 Nov 2022 22:33:43 GMT
Call-ID: 66FB517D-5FB511ED-AB5FFC01-A571E559@192.168.22.10
User-Agent: Cisco-SIPGateway/IOS-15.6.3.M9
Max-Forwards: 70
Timestamp: 1668033236
CSeq: 102 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=611,OS=97760,PR=441,OR=70560,PL=0,JI=0,LA=0,DU=4
Session-ID: 7ec3812c5ead52f5adabc7d00cde9275;remote=ef729addfbd05acf8c190cdb88602e99
Content-Length: 0


VoIP-Secoli#
Nov 9 22:33:56.274: //8427/66F97C6CAB59/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.10:5060;branch=z9hG4bK9551180;received=200.170.192.130;rport=57236
From: "teste" <sip:115555@192.168.22.10>;tag=50B32D4-51A
To: <sip:1155551144334433@3.3.3.3>;tag=as6f66231f
Call-ID: 66FB517D-5FB511ED-AB5FFC01-A571E559@192.168.22.10
CSeq: 102 BYE
Server: BR-SAO-CO1-SVV-SBC-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

 

On the changed translation not working. In your first post you mentioned that the translation was to replace the leading 0 (zero) with 115555, but now in you last response and in you debugs it shows that you do this translation, replace leading 0 (zero) with 11555511. With this i mind you would need to do this to make your rule a little better formatted.

voice translation-rule 1
 rule 1 /^0\(.*\)/ /11555511\1/

As you also have other rules I would suggest that you do  similar clean-up for these.

Apart from this you have an overlap in your configuration that negates the first occurrence of it.

*First one*
voice translation-profile CALL-IN
translate called 1

*Second one*
voice translation-profile CALL-IN
translate called 2

Also in general you can simplify you configuration by doing this.

voice translation-rule 1 ! Combine all destination patterns into one rule and put them in the proper order to translate correctly
 rule 1 /^00\([38]00\)/ /115555\1/
 rule 2 /^0\(00.*\)/ /115555\1/
 rule 3 /^00\(.*\)/ /115555\1/
 rule 4 /^0\(.*\)/ /11555511\1/
!
voice translation-profile PSTN-OUT ! New translation profile to be use for all egress calls to PSTN
 translate called 1
!
voice class e164-pattern-map 2000 ! Combine all destination patterns into one map and use this on a single outbound dial peer
 description E164 Pattern Map for called number to PSTN
  e164 00[38]00T
  e164 000[1-9]T
  e164 00[1-9]T
  e164 0[1-9]T
!
dial-peer voice 1 voip
 no session target ipv4:3.3.3.3 ! Not needed as it is used for outbound dial peers
!
dial-peer voice 20 voip
 description Ligacao call to PSTN
 translation-profile outgoing PSTN-OUT ! Use new translation profile
 no destination-pattern 0[1-9]T ! Not needed with the use of a map
 destination e164-pattern-map 2000 ! Use the combined map for all egress calls
 voice-class sip bind control source-interface GigabitEthernet0/1 ! Add missing bind statement
 no clid network-number 115555 ! Should not be needed and if it is needed to send calling number as 115555 all the time it can be added back or replaced by a translation rule that translate the calling number
!
no dial-peer voice 21 voip ! Not needed with all calls egressing out via dial peer 20
!
no dial-peer voice 22 voip ! Not needed with all calls egressing out via dial peer 20
!
no dial-peer voice 23 voip ! Not needed with all calls egressing out via dial peer 20
!
no dial-peer voice 24 voip ! Not needed with all calls egressing out via dial peer 20


voice service voip
 sip
  bind control source-interface GigabitEthernet0/0 ! Add bind staement that are needed for CME SIP phones
  bind media source-interface GigabitEthernet0/0 ! Add bind staement that are needed for CME SIP phones

I'll get back to you on the debugs once I've had time to look at them in detail.



Response Signature


Based on your output of the debugs you're not sharing your actual, or at least not the full configuration, and it does not give a clue to why you would just see a 0 (zero) on the phone.

This is what I see in your ccapi debug.
image.png
As you can see it uses dial peer 33 as the outbound and that's not shown in your previously shared information.

Please share your actual/complete running configuration, in an attached file and the output of the two before mentioned debugs and also debug voip translation, all running at the same time, as well in an attached file so that we can see what's going on in your gateway.



Response Signature