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
!
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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
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aaa new-model
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aaa authorization exec default local
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aaa session-id common
clock timezone AST -4 0
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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
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login on-success log
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subscriber templating
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multilink bundle-name authenticated
!
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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
!
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voice class uri 10000 sip
host ipv4:68.65.65.55 <----- Pointing to PSTN
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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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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
!
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control-plane
!
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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
!
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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
