Deploying Local Gateway for Webex Calling Problems

jorge-villa2 — Sat, 27 Sep 2025 18:07:20 GMT

Hello Community,

I need some assistance. I am implementing a Local Gateway (LGW) for Webex Calling. At the moment, I only have one tenant (9998) working properly. However, I am experiencing issues only with inbound calls.

When I try to receive calls for tenant 203, the traffic always matches dial-peer 99998, which belongs to tenant 9998, instead of hitting the correct inbound dial-peer for tenant 203. Outbound calls are working without any issues.

Has anyone experienced a similar problem or can provide recommendations on how to refine the dial-peer configuration to ensure the correct inbound call routing? Any guidance would be greatly appreciated.

Cisco advised me that the dial-peer labeling/structure should follow a specific format (as shown in the attached screenshot).

Building configuration...

Current configuration : 18007 bytes
!

hostname Webex-LGW-1
!
boot-start-marker
boot-end-marker
!
!
vrf definition VOIP_TRUNK_INTERNET
description PSTN
rd 1:400
!
address-family ipv4
route-target export 1:400
route-target import 1:400
exit-address-family
!
aaa new-model
!
!
aaa authorization exec default local
!
!
aaa session-id common
clock timezone AST -4 0
!
!
!
!
!
!
!
!
!
!
!
!
ip name-server 1.1.1.3 208.67.222.222
ip name-server vrf VOIP_TRUNK_INTERNET 1.1.1.3 208.67.222.222
ip domain name test.local
!
!
!
login on-success log
!
!
subscriber templating
!
!
!
!
!
!
multilink bundle-name authenticated
!
!
!
!
!
!

!
!
!
!
!
!
voice service voip
ip address trusted list
Ommited all ipv4 ip are configure
media statistics
media bulk-stats
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
trace
stun
stun flowdata agent-id 1 boot-count 16
stun flowdata shared-secret 6 \`D[Pi[NCTefHXMYQFO_ScI_ObGXQK_\QdGEbhQ
sip
registrar server
early-offer forced
midcall-signaling passthru
g729 annexb-all
sip-profiles inbound
!
!
voice class uri 10000 sip
host ipv4:68.65.65.55 <----- Pointing to PSTN
!
voice class uri Customer203 sip
pattern dtg=homologacion-piso61087204843_lgu
!
voice class uri Customer9998 sip
pattern dtg=test-piso6-gw10314924087_lgu
voice class codec 1000
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
voice class stun-usage 1000
stun usage firewall-traversal flowdata
stun usage ice lite
!
!
voice class sip-profiles 203
rule 9 request ANY sip-header SIP-Req-URI modify "sips:(.*)" "sip:\1"
rule 10 request ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 11 request ANY sip-header From modify "<sips:(.*)" "<sip:\1"
rule 12 request ANY sip-header Contact modify "<sips:(.*)>" "<sip:\1;transport=tls>"
rule 13 response ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 14 response ANY sip-header From modify "<sips:(.*)" "<sip:\1"
rule 15 response ANY sip-header Contact modify "<sips:(.*)" "<sip:\1"
rule 20 request ANY sip-header From modify ">" ";otg=homologacion-piso61087204843_lgu>"
rule 30 request ANY sip-header P-Asserted-Identity modify "sips:(.*)" "sip:\1"
!
voice class sip-profiles 20
response ANY sip-header Contact modify "192.168.11.25" "65.56.40.55"
request ANY sip-header Contact modify "192.168.11.25" "65.56.40.55"
response ANY sip-header From modify "192.168.11.25" "65.56.40.55"
request ANY sip-header From modify "192.168.11.25" "65.56.40.55"
response ANY sip-header Via modify "192.168.11.25" "65.56.40.55"
request ANY sip-header Via modify "192.168.11.25" "65.56.40.55"
response ANY sdp-header Audio-Connection-Info modify "192.168.11.25" "65.56.40.55"
request ANY sdp-header Connection-Info modify "192.168.11.25" "65.56.40.55"
response ANY sdp-header Connection-Info modify "192.168.11.25" "65.56.40.55"
request ANY sdp-header Session-Owner modify "192.168.11.25" "65.56.40.55"
response ANY sdp-header Session-Owner modify "192.168.11.25" "65.56.40.55"
!
voice class sip-profiles 10
response ANY sip-header Contact modify "65.56.40.55" "192.168.11.25"
request ANY sip-header Contact modify "65.56.40.55" "192.168.11.25"
request ANY sip-header SIP-Req-URI modify "65.56.40.55" "192.168.11.25"
request ANY sip-header SIP-Req-URI modify "65.56.40.55" "192.168.11.25"
request ANY sip-header SIP-Req-URI modify "65.56.40.55" "192.168.11.25"
response ANY sdp-header Audio-Connection-Info modify "65.56.40.55" "192.168.11.25"
response ANY sdp-header Connection-Info modify "65.56.40.55" "192.168.11.25"
request ANY sdp-header Audio-Connection-Info modify "65.56.40.55" "192.168.11.25"
request ANY sdp-header Connection-Info modify "65.56.40.55" "192.168.11.25"
!
voice class sip-profiles 9998
rule 9 request ANY sip-header SIP-Req-URI modify "sips:(.*)" "sip:\1"
rule 10 request ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 11 request ANY sip-header From modify "<sips:(.*)" "<sip:\1"
rule 12 request ANY sip-header Contact modify "<sips:(.*)>" "<sip:\1;transport=tls>"
rule 13 response ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 14 response ANY sip-header From modify "<sips:(.*)" "<sip:\1"
rule 15 response ANY sip-header Contact modify "<sips:(.*)" "<sip:\1"
rule 20 request ANY sip-header From modify ">" ";otg=test-piso6-gw10314924087_lgu>"
rule 30 request ANY sip-header P-Asserted-Identity modify "sips:(.*)" "sip:\1"
!
!
!
voice class dpg 10000
description PSTN
dial-peer 10000 preference 1
!
voice class dpg 203
description Incoming PSTN (DP9203) to WxC(DP203)
dial-peer 203 preference 1
!
voice class dpg 9998
description Incoming PSTN (DP99998) to WxC(DP9998)
dial-peer 9998 preference 1
!
voice class tenant 10000
no connection-reuse
session transport udp
url sip
error-passthru
bind control source-interface GigabitEthernet2
bind media source-interface GigabitEthernet2
no pass-thru content custom-sdp
sip-profiles 20
sip-profiles 10 inbound
!
voice class tenant 203
tls-profile 1
listen-port secure 5064
registrar dns:us10.bcld.webex.com scheme sips expires 240 refresh-ratio 50 tcp tls
credentials number Homologacion-Piso60241171077_LGU username Homologacion-Piso61087204843_LGU password 6 NOSPPS^PgJH[FCdUCdIbL[GARWN\PaAiP]NP realm BroadWorks
authentication username Homologacion-Piso61087204843_LGU password 6 `^ff]VFVfIdRFiHCTKXbNV[]DYWTQUR^INDi realm BroadWorks
authentication username Homologacion-Piso61087204843_LGU password 6 UIYPc[DXeP^MCaN\bZcYb[fFChE\bNeL`MaU realm us10.bcld.webex.com
no remote-party-id
sip-server dns:us10.bcld.webex.com
connection-reuse
srtp-crypto 1000
session transport tcp tls
url sips
error-passthru
asserted-id pai
bind control source-interface GigabitEthernet1
bind media source-interface GigabitEthernet1
no pass-thru content custom-sdp
sip-profiles 203
outbound-proxy dns:jfk05.sipconnect-us.bcld.webex.com
privacy-policy passthru
!
voice class tenant 9999
bind control source-interface GigabitEthernet2
bind media source-interface GigabitEthernet2
no pass-thru content custom-sdp
sip-profiles 20
sip-profiles 10 inbound
!
voice class tenant 9998
tls-profile 1
listen-port secure 5065
registrar dns:us10.bcld.webex.com scheme sips expires 240 refresh-ratio 50 tcp tls
credentials number test-Piso6-GW10014302153_LGU username test-Piso6-GW10314924087_LGU password 6 PR[PFZ]LdDFAcJSCX[MeiJIbVdbVRN[AZTa^ realm BroadWorks
authentication username test-Piso6-GW10314924087_LGU password 6 TeUbgMK_iBRQUVMfL\AbSAFUC\gPO_ANLSX` realm BroadWorks
authentication username test-Piso6-GW10314924087_LGU password 6 LPX\XKThUgTWJX^]SLJDHeeB_]aK[KDdKOMg realm us10.bcld.webex.com
no remote-party-id
sip-server dns:us10.bcld.webex.com
connection-reuse
srtp-crypto 1000
session transport tcp tls
url sips
error-passthru
asserted-id pai
bind control source-interface GigabitEthernet1
bind media source-interface GigabitEthernet1
no pass-thru content custom-sdp
sip-profiles 9998
outbound-proxy dns:jfk05.sipconnect-us.bcld.webex.com
privacy-policy passthru
!
voice class srtp-crypto 1000
crypto 1 AES_CM_128_HMAC_SHA1_80
!
voice class tls-profile 1
trustpoint dummyp
!
voice class tls-profile 2
trustpoint dummyp
!
!
!
!
voice translation-rule 203
rule 1 /1000/ /+17875214000/
!
voice translation-rule 9998
rule 1 /1000/ /+17875224444/
rule 2 /1001/ /+17875224565/
!
!
voice translation-profile 203
translate called 203
!
voice translation-profile 9998
translate called 9998
!
!
!
!
!
!
!
!
!
!
!
!
!
!
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!
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!
!
!
!
!
!
!
!
interface GigabitEthernet1
description to Webex
ip address 192.168.10.25 255.255.255.0
no ip redirects
no ip proxy-arp
negotiation auto
no mop enabled
no mop sysid
!
interface GigabitEthernet2
description Traffic to PSTN 65.56.40.55
vrf forwarding VOIP_TRUNK_INTERNET
ip address 192.168.11.25 255.255.255.0
no ip redirects
no ip proxy-arp
negotiation auto
no mop enabled
no mop sysid
!
ip forward-protocol nd
!
ip http server
ip http authentication local
ip http secure-server
ip route 0.0.0.0 0.0.0.0 192.168.10.1
ip route vrf VOIP_TRUNK_INTERNET 0.0.0.0 0.0.0.0 192.168.11.1
!
!
!
!
!
!
!
!
control-plane
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dial-peer voice 10000 voip
description Outbound IP PSTN Trunk
destination-pattern BAD.BAD
session protocol sipv2
session target ipv4:65.23.217.140
voice-class codec 1000
voice-class sip profiles 10 inbound
voice-class sip tenant 10000
dtmf-relay rtp-nte
no vad
!
dial-peer voice 203 voip
description Inbound/Outbound Webex Calling
max-conn 250
destination-pattern BAD.BAD
session protocol sipv2
session target sip-server
destination dpg 10000
incoming uri request Customer203
voice-class codec 1000
voice-class stun-usage 1000
no voice-class sip localhost
voice-class sip tenant 203
dtmf-relay rtp-nte
srtp
no vad
!
dial-peer voice 9998 voip
description Inbound/Outbound Webex Calling
max-conn 30
destination-pattern BAD.BAD
session protocol sipv2
session target sip-server
destination dpg 10000
incoming uri request Customer9998
voice-class codec 1000
voice-class stun-usage 1000
no voice-class sip localhost
voice-class sip tenant 9998
dtmf-relay rtp-nte
srtp
no vad
!
dial-peer voice 99998 voip
description Incoming dial-peer from PSTN
translation-profile incoming 9998
session protocol sipv2
destination dpg 9998
incoming uri via 10000
voice-class codec 1000
voice-class stun-usage 1000
voice-class sip tenant 9999
dtmf-relay rtp-nte
no vad
!
dial-peer voice 9203 voip
description Incoming dial-peer from PSTN
translation-profile incoming 203
session protocol sipv2
destination dpg 203
incoming uri request Customer203
incoming uri via 10000
voice-class codec 1000
voice-class stun-usage 1000
voice-class sip tenant 9999
dtmf-relay rtp-nte
no vad
!
!
sip-ua
transport tcp tls v1.2
crypto signaling default trustpoint testTp cn-san-validate server
tcp-retry 1000
!
!
line con 0
exec-timeout 0 0
logging synchronous
stopbits 1
line aux 0
line vty 0 4
transport input ssh
!
call-home
! If contact email address in call-home is configured as sch-smart-licensing@cisco.com
! the email address configured in Cisco Smart License Portal will be used as contact email address to send SCH notifications.
contact-email-addr sch-smart-licensing@cisco.com
profile "CiscoTAC-1"
active
destination transport-method http
ntp server time.google.com
ntp server 1.1.1.1
ntp server time.cloudflare.com
ntp server time.facebook.com
!
!
!
!
!
!
end

Re: Deploying Local Gateway for Webex Calling Problems

M02@rt37 — Sun, 28 Sep 2025 11:02:25 GMT

Hello @jorge-villa2

Please provide debug ccsip messages output during a call attempt.

Thanks.

Re: Deploying Local Gateway for Webex Calling Problems

jorge-villa2 — Sun, 28 Sep 2025 23:17:01 GMT

Problem that i having is always hit dial-peer 9203 when i call to dial-peer 99998. Both have URI but 9203 always win.

Sep 28 23:06:20.636: //-1/94A187EE80D6/CCAPI/cc_api_call_setup_ind_common:
Interface=0x7E57F35858F8, Call Info(
Calling Number=+1787556565,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=+17875223212(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=9203, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=377

Re: Deploying Local Gateway for Webex Calling Problems

jorge-villa2 — Mon, 29 Sep 2025 02:05:19 GMT

Hi Team,

I’m troubleshooting an issue where calls routed to dial-peer 99998 are failing, while calls routed to dial-peer 9203 are working as expected.

I’ve attached the debug outputs for reference (was unable to upload the .txt file since the system blocked it).

Thanks in advance for your time and support.

topic Deploying Local Gateway for Webex Calling Problems in Webex Calling

Deploying Local Gateway for Webex Calling Problems

Re: Deploying Local Gateway for Webex Calling Problems

Re: Deploying Local Gateway for Webex Calling Problems

Re: Deploying Local Gateway for Webex Calling Problems