Telekom SIP Trunk / DE
Dienstag, 22. September 2020 08:58
ip name-server 217.0.43.145 217.0.43.129
!
voice call send-alert
no voice call carrier capacity active
voice rtp send-recv
!
voice service voip
ip address trusted list
ipv4 217.0.0.0 255.255.0.0
ipv4 [CUCM Pub IP] 255.255.255.255
ipv4 [CUCM Sub IP] 255.255.255.255
rtp-port range 16384 32766
address-hiding
mode border-element license capacity [Session count]
media statistics
media bulk-stats
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
sip
rel1xx disable
session refresh
header-passing
error-passthru
asserted-id pai
conn-reuse
no update-callerid
midcall-signaling passthru
privacy-policy passthru
call-route p-called-party-id
pass-thru headers unsupp
sip-profiles inbound
audio forced
!
voice class uri 1000 sip
host ipv4:[CUCM Pub IP]
host ipv4:[CUCM Sub IP]
!
voice class codec 1000
codec preference 1 g711alaw
!
voice class sip-profiles 2001
rule 1 request INVITE sip-header P-Asserted-Identity remove
rule 2 request REINVITE sip-header P-Asserted-Identity remove
!
voice class sip-profiles 2000
rule 1 request INVITE peer-header sip P-Asserted-Identity copy "sip:(.*)@" u01
rule 2 request INVITE peer-header sip Diversion copy "sip:(.*)@" u01
rule 3 request INVITE sip-header P-Asserted-Identity modify "sip:.*@(.*)" "sip:\u01@\1"
rule 4 request ANY sip-header Min-SE remove
rule 5 request ANY sip-header Diversion remove
rule 6 request ANY sdp-header Connection-Info remove
rule 7 request ANY sip-header User-Agent remove (optional)
rule 8 response ANY sip-header User-Agent remove (optional)
rule 9 request ANY sip-header Cisco-Guid remove (optional)
rule 10 response ANY sip-header Cisco-Guid remove (optional)
!
voice class sip-profiles 3000
rule 1 request REGISTER sip-header Contact modify "<.*:.*@(.*)>" "<sip:\1;bnc>"
rule 2 request REGISTER sip-header Proxy-Require add "Proxy-Require: gin"
rule 3 request REGISTER sip-header Require add "Require: gin"
rule 4 request ANY sip-header User-Agent remove (optional)
rule 5 response ANY sip-header User-Agent remove (optional)
rule 6 request ANY sip-header Cisco-Guid remove (optional)
rule 7 response ANY sip-header Cisco-Guid remove (optional)
!
voice class sip-copylist 1000
sip-header P-Asserted-Identity
sip-header Diversion
!
voice class e164-pattern-map 1000
e164 11[68]T
e164 11[025]
e164 +T
e164 0T
!
GW Seite 1
!
voice class e164-pattern-map 2000
e164 +49[CLIENT PUBLIC NUMBER]T
!
voice class server-group 1000
ipv4 [CUCM PUB IP] preference 2
ipv4 [CUCM SUB IP] preference 1
description ### CUCM Server Group ###
!
voice class sip-options-keepalive 1000
up-interval 30
retry 3
!
voice class tenant 1000
no remote-party-id
timers buffer-invite 5000
session transport tcp
session refresh
header-passing
error-passthru
bind control source-interface [LAN INTERFACE]
bind media source-interface [LAN INTERFACE]
no pass-thru content custom-sdp
privacy-policy passthru
!
voice class tenant 2000
registrar dns:sip-trunk.telekom.de expires 240 tcp auth-realm sip-trunk.telekom.de
credentials number [TELEKOM REGISTRATION NUMBER] username [TELEKOM REGISTRATION ID] password [TELEKOM REGISTRATION PASSWORD] realm sip-trunk.telekom.de
authentication username [TELEKOM REGISTRATION ID] password [TELEKOM REGISTRATION PASSWORD]
no remote-party-id
timers buffer-invite 5000
timers dns registrar-cache ttl
sip-server dns:sip-trunk.telekom.de
session transport tcp
session refresh
no update-callerid
header-passing
error-passthru
bind control source-interface [WAN INTERFACE]
bind media source-interface [WAN INTERFACE]
asserted-id pai
no pass-thru content custom-sdp
conn-reuse
sip-profiles 3000
outbound-proxy dns:reg.sip-trunk.telekom.de
privacy-policy passthru
!
dial-peer voice 1000 voip
description ### From / To CUCM ###
huntstop
session protocol sipv2
session server-group 1000
destination e164-pattern-map 2000
incoming uri via 1000
voice-class codec 1000
voice-class sip tenant 1000
voice-class sip block 183 sdp present
voice-class sip options-keepalive profile 1000
voice-class sip copy-list 1000
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate 14400
fax nsf 000000
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
no vad
!
dial-peer voice 2000 voip
description ### From / To SIP-Provider DTAG ###
huntstop
session protocol sipv2
session target sip-server
destination e164-pattern-map 1000
incoming called e164-pattern-map 2000
voice-class codec 1000
voice-class sip profiles 2000
voice-class sip profiles 2001 inbound
voice-class sip tenant 2000
voice-class sip block 183 sdp present
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate 14400
fax nsf 000000
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
GW Seite 2
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
ip qos dscp cs6 signaling
clid strip name
no vad
Access Liste for WAN-Interface
ip access-list extended FROM-DT-TO-CUBE
remark ### Permitted ISP Public address Range ###
permit udp 217.0.0.0 0.0.255.255 range 1025 65534 any
permit tcp 217.0.0.0 0.0.255.255 any eq 5060
permit tcp 217.0.0.0 0.0.255.255 any established
remark ### Deutsch Telekom DNS ###
permit udp host 217.0.43.145 eq domain any
permit udp host 217.0.43.129 eq domain any
remark ### Permitted PING from inside to outside ###
permit icmp any any echo-reply
remark ### Deny all Other traffic ###
deny ip any any
!
interface [WAN-Interface]
ip access-group FROM-DT-TO-CUBE in
GW Seite 3
CUCM Konfiguration
Dienstag, 22. September 2020 09:09
SIP Profile:
--> Copy of Standard SIP Profile
• SIP Rel1xx Options: Send PRAK for All 1xx Messages
• Early Offer Support for Voice and Video Calls: Best Effort (No MTP inserted)
• SIP Options Ping: On (optional)
• SDP Information: Send Send-receive SDP in mid-Call Invite
SIP Trunk Security Profile:
--> Copy of Non Secure SIP Trunk Profile
SIP Trunk:
PSTN Access
Run on all active Unified CM Nodes
Remote Party id: OFF
Asserted-Identity: ON
Asserted-Type: PAI
Redirecting Diversion Header Delivery - Outbound
SIP Trunk Security Profile: select the added on from above
SIP Profile: select the added on from above
GW Seite 1
Service Parameter:
--> CUCM-Node --> Service "Cisco Callmanager":
• Duplex Streaming Enabled => True
--> CUCM-Node --> Service "Cisco IP Voice Media Streaming App":
• Supported MOH Codecs: 711 alaw should be marked
GW Seite 2
GW Seite 3
CUC Konfiguration
Donnerstag, 14. Januar 2021 09:48
Telephony Integrations:
--> Check Port Group, if in "G.711a-law" is select in "Codec Advertising"
GW Seite 1
