4.10 Voice Call Routing VRT (Voice Routing Table)

4.10 Voice Call Routing VRT (Voice Routing Table)

Copyright © OneAccess Networks – All rights reserved
VoIP: Call routing Overview Applicable to any call between any type of ports (local or voip). Any call routing is possible except VoIP -> VoIP call Implicit Routing Easy to configure : no called number configuration No number processing Routing Table Incoming call routing : optional process done for each incoming call (from VoIP network or local port) allowing number translations (calling & called numbers) and voice profile selection Outgoing call routing : for numbering plan definition (overlap or in-bloc), selection of the remote voip device if no gatekeeper, number translations (calling & called) Possibility to configure a backup route in case of outgoing call failure Possibility to modify specific ISDN Information Elements : Type of number, Numbering plan, Bearer capability Copyright © OneAccess Networks – All rights reserved

Call Routing Architecture (from local port)
Incoming call routing (number translations) Outgoing call routing PSTN Interfaces (ISDN, analog) VoIP Network (SDP media properties) Copyright © OneAccess Networks – All rights reserved

Call Routing Architecture (from VoIP network)
Incoming call routing (select voip profile, number translations) Outgoing call routing PSTN Interfaces (ISDN, analog) VoIP Network (SDP media properties) Copyright © OneAccess Networks – All rights reserved

VoIP: Call Routing Up to 200 routing rules can be configured Type of rule: incoming (initial routing when receiving the call) or outgoing (2nd step of routing) Number to be checked Called or calling number Wildcards can be used Related port (local or VoIP) Actions: Number translations, ISDN I.E. forcing Definition of backup routes The rule order is important: the rule #1 is checked first Insertion & deletion of rules by using the rule index Copyright © OneAccess Networks – All rights reserved

VoIP: Call Routing Routing Process for call coming from a local port 1. The incoming rules related to the local port are checked down the rule list If matching the rule (number & port), the number translations are applied The check of incoming routing rules can be processed again until the last one. This possibility allows to make several number translations on the calling and called number. 2. The outgoing rules are checked from the first one If fully matching the rule, the number translations of the rule are applied and the call is routed to the specified port (local or voip). Backup route can be configured to define alternative rules in case of failure. If partially matching the rule (e.g.: incomplete number in overlap mode), the call is not further routed and the GW waits for another digit Routing process summary Case of a call coming from a local port The routing has the following general structure: incoming call routing and transformation Translation Routing Outgoing routing 1. The calling number translation defined in the dial-peer voice pots is applied. 2. The routing table is checked against the incoming routing rules (with prefix-type = incoming or in-andout) according to the configured number type (calling or called) and with the dial-peer = pots-group corresponding to the local port. 3. If a match is found, translations on calling and/or called numbers are applied. If ‘next’ is configured, the routing table is checked again from the specified rule with the translated numbers. If a match is found, translations on calling & called numbers are applied again. This process is applied until the end of the table or until a "last" option is encountered. This process enables several number translations. 4. The implicit-routing parameter of the dial-peer pots is checked. If defined, the call is directly routed to the corresponding VoIP peer (dial-peer voip) or local port (pots). The routing process is completed. 5. The routing table is checked (from the first rule) against the outgoing rules (prefix-type = outgoing or inand- out) according to the number type (calling or called). 6. Upon a total match, translations on calling and/or called numbers are applied and the call is routed to the dial-peer voip port. The routing process is finished. It will be restarted from this step in case of backup (and go to step 5 from the following routing rule). 7. Upon a partial match (in case of overlap dialing only), the routing is considered inconclusive but the endof- dialing timer is started. The routing process will be done again with the following digit or upon the endof- dialing timer expiry 8. If no match, the call is cleared. Copyright © OneAccess Networks – All rights reserved

VoIP: Call Routing Routing Process for call coming from VoIP 1. The incoming rules related to voip are checked from the first one Upon a match (number only), the number translations are applied The voip port (coder profile,etc) is determined Check of routing table can be processed again until the end of the table 2. The outgoing rules are checked from the first one Outgoing rules with voip port are excluded (no voip-voip routing) Upon a full match, the number translations are applied and the call is routed to the specified port (local or voip). Backup can be configured to define alternative rules in case of failure. Upon a partial match (ex : uncomplete number in overlap mode), the routing process is fully cancelled (done again with the next digit). Routing process summary Case of a call coming from the H.323/SIP Network 1. The routing table is checked against the incoming routing rules (with prefix-type = incoming or in-andout) according to the configured number type (calling or called) and with the dial-peer = voip. In this step, the dial-per voip identifier is unknown. 2. Upon a match, the dial-peer voip identifier is taken into account to determine for example a voice coding profile. Translations on calling and/or called numbers are applied. If "next» is configured, the routing table is checked again from the specified rule with the translated numbers. Upon a match, translations on calling&called numbers are applied again. This process is applied until the end of the table or until a "last" option is encountered. Then go directly to step 4 3. If no match, the source IP address is compared to the address configured in all the dial-peer voip. In case of match, the corresponding dial-peer voip is selected. If not, the lowest dial-peer voip identifier is selected by default. 4. The implicit-routing parameter of the selected dial-peer voip is checked. If defined, the call is directly routed to the corresponding pots group. The routing process is finished. 5. The routing table is checked (from the first rule) against the outgoing rules (prefix-type = outgoing or inand- out) according to the number type (calling or called) and with dial-peer = pots-group. 6. Upon a total match, translations on calling and/or called numbers are applied and the call is routed to the dial-peer pots group. The routing process is finished. It will be restarted from this step in case of backup (and go to step 13 from the following routing rule). O N E O S V R 1 1 V O I C E U S E R G U I D E Page of 129 7. Upon a partial match (in case of overlap dialing only), the routing is considered inconclusive and the endof- dialing timer is optionally started. All the routing process will be done again with the fully received number when the following digit will be received. Upon the end-of-dialing timer expiry, the last outgoing routing with a partial match is taken into account. 8. If no match, the call is cleared. Copyright © OneAccess Networks – All rights reserved

VoIP: Call Routing Routing on local port Configuration of a port group Same number for several ports Incoming Calls are distributed with a configurable priority Possibility of call hunting automatic backup inside the port group higher priority is given to « working » ports Several numbers can be configured for each port Copyright © OneAccess Networks – All rights reserved

Voice Routing (1) 6. Voice routing one200(configure)>voice-routing one200(voice-route)> ? display-routes Show voice routing table exit Exit from command node insert Insert VOIP voice route move Move VOIP voice route no no route Set VOIP voice route test-route Test voice routing table <cr> one200(voice-route)> Copyright © OneAccess Networks – All rights reserved

Voice Routing (2) one200(voice-route)>route 1 one200(conf-voice-route)> ? dial-peer Set route dial peer exit Exit from command node force-bearer-cap Set force bearer capability fields force-clir Set caller line identity request force-numplan Set origin/destination numplan insert-calling Set route calling insertion insert-prefix Set route prefix insertion insert-suffix Set route suffix insertion no no prefix Set route prefix(prefix [number-type][length][timer] [overlap]) prefix-type Set route direction and type of call startup-file Set restart equipment with the file name. suppress-prefix Set route prefix suppression translate Set route number translate wildcard Set wildcard value for prefix <cr> Copyright © OneAccess Networks – All rights reserved

Voice Routing (3) Registration to registrar server (user=phone). E164 Prefix is mapped to a pots-group or a voip resource. To register it, <ua-sip> must be added. Corresponding prefix is used as user portion of sip uri of To/From/Contact header field To register a username instead of a E164 prefix (no user=phone), sip- username must be validated. It overrides « prefix » value for To/From/Contact header fields for register method. It overrides « prefix » value for From/Contact header fields for Invite method. Registrar ip address or domain name must be configured at the sip- gateway (reg-dns-add) 401 or 407 challenge for register and/or invite method. Sip- authentication <username> <password> are provided to calculate MD5 response. Copyright © OneAccess Networks – All rights reserved

Voice Routing (4) authentication CLI(voice-route)>route 1 CLI(voice-route)>dial-peer pots-group 0 ua-sip CLI(voice-route)>sip-username <user portion of sip uri> CLI(voice-route)>sip-authentication [<username> <password>] CLI(voice-route)>prefix 1000 length 4 CLI(voice-route)>prefix-type outgoing called last CLI(voice-route)>exit CLI(configure)>exit Copyright © OneAccess Networks – All rights reserved

Voice Routing (5) one200(voice-route)>display-routes all length 4 / voip 1 / mixed - called - next all length 4 / pots 0 / outgoing - called - last one200>show running-config ... voice-routing route 1 dial-peer voip 1 prefix 20. length 4 prefix-type mixed called next exit route 2 dial-peer pots-group 1 ua-sip prefix length 4 prefix-type outgoing called last Copyright © OneAccess Networks – All rights reserved

Voice Routing (6) Prefix suppression 1234 Pots-group 0 VoIP 0 route 1 dial-peer pots-group 0 prefix-type outgoing called last prefix 120A. length 4 suppress-prefix 4 called wildcard A 0123 exit Copyright © OneAccess Networks – All rights reserved

Voice Routing (7) Codec Profile for incoming call Pots-group 0 Incoming Table VoIP 1 VoIP 2 Pots-group 4 route 14 dial-peer voip 1 prefix length 9 prefix-type incoming called exit route 15 dial-peer voip 2 prefix . length 0 Copyright © OneAccess Networks – All rights reserved

Voice Routing (8) Modification of Calling and Called numbers 1/4 Calling : Calling : Called : Incoming Table Outgoing Table Called : Calling : Called : Calling : Called : Pots-group 0 Copyright © OneAccess Networks – All rights reserved

Voice Routing (9) Modification of Calling and Called numbers 2/4 route 2 dial-peer pots-group 0 translate A A calling prefix . length 0 prefix-type incoming calling next exit route 3 dial-peer pots-group 0 alias insert-prefix 1200 called prefix length 9 prefix-type incoming calling alias added to register the number 1200 in the RAS Copyright © OneAccess Networks – All rights reserved

Voice Routing (10) Modification of Calling and Called numbers 3/4 Calling : Called : Calling : Calling : Called : Incoming Table Outgoing Table Called : Voip 1 Calling : Called : Calling : Called : Voip 0 Pots-group 0 Calling : Called : Copyright © OneAccess Networks – All rights reserved

Voice Routing (11) Modification of Calling and Called numbers 4/4 route 4 dial-peer voip 1 prefix length 4 prefix-type outgoing called last exit route 5 dial-peer voip 0 prefix . length 0 Copyright © OneAccess Networks – All rights reserved

Voice Toubleshooting & Statistics

Voice routing table testing
One_Training(voice-route)>display-route 1 - . all length 10 / pots 0 / incoming - called - last / +prefix-calling 0 all timer / voip 0 / outgoing - called - last all length 9 / pots 0 / outgoing - called - last / -prefix-called 1 One_Training(voice-route)> One_Training> configure terminal One_Training(configure)>voice-routing One_Training(voice-route)>test-route from-voip 0 --> test_route : Calling= Called= Route match : Calling= Called= Incoming Routes = Outgoing Routes = 20, Incoming call from voip id:0 Send towards Isdn on local port: 5/0 One_Training(voice-route)>test-route from-pots 0 --> test_route : Calling= Called= Route match : Calling= Called= Incoming Routes = 1, Outgoing Routes = 10, Outgoing call from local port: 5/0 -> Send towards H323 on voip id:0 Copyright © OneAccess Networks – All rights reserved

VoIP Statistics BRI statistics CLI# show voice voice-port bri index 0 voice port /0 protocol descriptor BRI_NT current state activated config state up layer 1 status activated attached vmoabri dial peer number of voice communication 0 bri Tx frames on D channel bri Rx frames on D channel Outgoing calls : 102 Outgoing calls failures : 5 Physical Interface down : 0 Cause Class 0 (normal event) : 0 Cause Class 1 (normal event) : 5 Normal Cause (16) : 2 User busy (17) : 3 No answer (18) : 0 Cause Class 2 (unavailable ressources) : 0 Cause Class 3 (unavailable service) : 0 Cause Class 4 (service not provided) : 0 Cause Class 5 (invalid message) : 0 Cause Class 6 (protocol error) : 0 Cause Class 7 (interworking) : 0 Incoming calls : 54 Incoming calls failures : 7 Remote failure : 0 Unknown number : 5 DSP unavailable : 0 Not specified : 2 Copyright © OneAccess Networks – All rights reserved

VoIP Statistics PRI statistics CLI# show voice voice-port pri index 0 voice port /0 physical type E1 protocol descriptor E1_PRI current state activated config state up layer 1 status deactivated number of voice communications 0 pri AIS occurence pri RDI occurence Outgoing calls : 67 Outgoing calls failures : 3 Physical Interface down : 0 Cause Class 0 (normal event) : 0 Cause Class 1 (normal event) : 3 Normal Cause (16) : 0 User busy (17) : 3 No answer (18) : 0 Cause Class 2 (unavailable ressources) : 0 Cause Class 3 (unavailable service) : 0 Cause Class 4 (service not provided) : 0 Cause Class 5 (invalid message) : 0 Cause Class 6 (protocol error) : 0 Cause Class 7 (interworking) : 0 Incoming calls : 23 Incoming calls failures : 2 Remote failure : 2 Unknown number : 0 DSP unavailable : 0 Not specified : 0 Copyright © OneAccess Networks – All rights reserved

VoIP Statistics FXS statistics CLI# show voice voice-port fxs index 0 voice port /0 current state on hook config state up attached vmoa fxs dial peer 0 voice communication no Outgoing calls : 32 Outgoing calls failures : 3 User busy : 2 No answer : 1 Incoming calls : 6 Incoming calls failures : 0 Remote failure : 0 Unknown number : 0 DSP unavailable : 0 Not specified : 0 Copyright © OneAccess Networks – All rights reserved

VoIP Statistics Dial-peer VoIP Statistics (1) one200> show voice dial-peer voice voip type index <port id> [reset] or one200> show voice dial-peer voice voip type global[reset] type may be : current : statistics on current calls outgoing : outgoing calls only incoming : incoming calls only user-plan : voice & fax only all (default) : all the statistics are provided Copyright © OneAccess Networks – All rights reserved

VoIP Statistics Dial-peer VoIP Statistics (2): Outgoing Calls Dial Peer Current Calls Outgoing Calls Outgoing Calls Outgoing calls failures RAS Call Failures Gatekeeper Unavailable Admission Rejects H225/Q931 Call failures Cause Class 0 (normal event) Cause Class 1 (normal event) Normal Cause (16) User busy (17) No answer (18) Cause Class 2 (unavailable ressources) 0 Cause Class 3 (unavailable service) 0 Cause Class 4 (service not provided) 0 Cause Class 5 (invalid message) Cause Class 6 (protocol error) Cause Class 7 (interworking) H245 Call failures Incompatible capabilities Protocol errors Internal call failures DSP unavailable Max-bandwidth exceeded Max-connection exceeded Not specified Copyright © OneAccess Networks – All rights reserved

VoIP Statistics Dial-peer VoIP Statistics (3): Incoming Calls Incoming calls Incoming calls failures RAS Call failures Gatekeeper Unavailable Admission Rejects Local Port Call failures H245 Call failures Incompatible capabilities Protocol errors Internal call failures DSP unavailable Unknown number Channel / port unavailable Max-bandwidth exceeded Max-connection exceeded Not specified Copyright © OneAccess Networks – All rights reserved

VoIP Statistics Dial-peer VoIP Statistics (4): Voice and Fax RTP statistics Number of transmitted packets Number of received packets Number of transmitted bytes Number of received bytes Number of excessive jitter events Number of lost packets Number of invalid packets Number of calls with frame error rate total <0.01<0.1<0.5<1<5>=5 Modem passthrough Number of switching to modem mode T38 FAX Calls Number of outgoing fax Number of incoming fax Number of failures Request Mode failure Pre-message procedure failure Page failure Number of transmitted packets Number of received packets Number of transmitted bytes Number of received bytes Copyright © OneAccess Networks – All rights reserved

VoIP Statistics Events vxTarget>event filter Add/remove events filters manager Add a SNMP manager no No recover Recover events from memory vxTarget>event filter add Add an event filter remove Remove a events filter from the table vxTarget>event filter add vox ALL All families from vox group GEN GEN VOATM VOATM VOIP VOIP vxTarget>event filter add vox voip <subfam> <ALL | ControlPlan | UserPlan> <fam2> <GEN | VOATM> vxTarget>event filter add vox voip all show Copyright © OneAccess Networks – All rights reserved

VoIP Statistics Voice call history, active calls Gives statistics on the current voice calls and the last 100 calls vxTarget>show voice voip-call any ind 1 1 - Call from remote voip: 0, to local port: 5/1 call-id: 4 active calling : 110, called : 111 setup time: 01/02/00 04h58m31s 01/02/00 04h58m31s RTP Source ip : rtp:16384 /Dest ip : rtp:16386 (active) Play time (voice) : 00h00m39s Tx Coder : G729 / 20 ms ; Rx Coder : G729 RTP Packets RX / TX : 1988 / 1989 RTP Packet lost&discarded RX / TX (RTCP reported) : 0 / Number of Excessive Jitter events : 3 Copyright © OneAccess Networks – All rights reserved

VoIP Statistics RTP sessions history Gives complete statistics about the 200 last RTP sessions CLI> show voice rtpcall full any ind 2 2 - 01/04/01 00h47m24s RTP :16384 – :16386 Play time (voice) : 00h00m46s Tx Coder : G729 / 20 ms ; Rx Coder : G729 VAD enabled local / remote : no / no ERL : 15 dB ACOM : 32 dB RTP Packets received (DSP / Uplink) : 2337 / 2337 lost : 0 out of sequence : 0 invalid : 0 RTP Packets transmitted (DSP / Uplink) : 2338 / 2338 lost (RTCP reported) : 0 Jitter parameter : 100 ms Number of Excessive Jitter events : 1 Copyright © OneAccess Networks – All rights reserved

VoIP Statistics RTP sessions history (continue) Excessive Jitter events : 2| 1| * 0 30" 1' 2' 4' 8' 12' >16' Jitter received (uplink) : Max delay : 93 ms Delays (ms) >50 >100 >150 >200 >300 Nb of occur Interarrival max jitter : 9 ms Jitter received (DSP) : Frames with a delay >50 ms : 1| * * Jitter transmitted (uplink) : Max delay : 6 ms Nb of occur Interarrival max jitter : 1 ms (RTCP reported) : 2 ms Copyright © OneAccess Networks – All rights reserved

VoIP: Internal Call Generator
Possibility to generate and / or terminate one or several VoIP calls Two services: RTP loopback or BERT testing Use of virtual and routable dial-peer pots dial-peer voice pots 4 service bert2047 both 3 pots-group 0 exit voip-call 1 pots 4 called calling 3000 bearer data duration 180 timeout 10 One200> start identifier 1 Copyright © OneAccess Networks – All rights reserved

VoIP: ISDN capture Possibility to capture the signalling traffic over ISDN BRI & PRI interfaces: layer 1 to 3 For VoIP side: use of the IP capture possibilities CLI>conf t CLI>logging buffered debug CLI>exit CLI>debug isdn all layer 1to3 00:07: line:5/0 L1 frame sent. 00:07: line:5/0 L2 tx UI P/F=0 NR=4 NS=2 C/R=1. 00:07: hex: 02 ff 03 00:07: line:5/0 L3 tx SETUP callref:8. 00:07: hex1: a a1 31 00:07: hex2: a1 00:07: line:5/0 L1 frame received. 00:07: line:5/0 L2 rx SABME P/F=1 C/R=0. 00:07: hex: f 00:07: line:5/0 L1 frame sent. 00:07: line:5/0 L2 tx UA P/F=1 NR=4 NS=2 C/R=0. 00:07: hex: Copyright © OneAccess Networks – All rights reserved

Call factory over IP For debug, a SETUP can be sent on VoIP. One_training>auto-call <called> called number: up to 21 characters <0..9, #, *> One_training>auto-call <calling> calling number: up to 21 characters <0..9, #, *> <pots-number> pots: 0..29 <bearer> bearer capability < voice | data | voiceband > overlap units in milliseconds: <0 means no overlap used> <cr> One_training>auto-call 17:50: Info vox factory test 1 call-id: 4, ident: auto-call, CALL IN PROGRESS Calling= 005 Called= one100_interopBW>17:50: Info vox voip controlplan 3 Incoming call on local pots: 0, calling: , called: , call-id: 4. 17:50: Info vox voip controlplan 3 Outgoing call on voip id: 0, calling: , called: , call-id: 4. 17:50: Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED cause=no codec. 17:50: Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED on pots cause=[Norma l call clearing]. Copyright © OneAccess Networks – All rights reserved

Auto call to ISDN For debug, a ‘SETUP’ can be sent on a ISDN local port. One_training>isdn test call ( data call/unrestricted ) 02:27: line:5/0 L1 event received PH_AR State:F3. 02:27: line:5/0 L1 event received EV_LOST_FRAMING State:F4. 02:27: line:5/0 L1 event received EV_INFO_2 State:F5. 02:27: line:5/0 L1 event received EV_INFO_4_8(PH_AI) State:F6. 02:27: line:5/0 L1 event received MPH_AI State:F7. 02:27: line:5/0 L1 frame sent. 02:27: line:5/0 L2 tx SABME P/F=1 C/R=0. 02:27: hex: f 02:27: line:5/0 L1 frame received. 02:27: line:5/0 L2 rx UA P/F=1 C/R=0. 02:27: hex: 02:27: line:5/0 L1 frame sent. 02:27: line:5/0 L2 tx INFO P=0 NR=0 NS=0 C/R=0. 02:27: hex: 02:27: line:5/0 L3 tx SETUP callref:4. 02:27: hex1: 02:27: hex2: a1 02:27: Called Number : 85841 Copyright © OneAccess Networks – All rights reserved

Fax relay T38 debug Fax T38 processing and traces: display of the T30 messages. One_training>trace filter add vox up ifp 2 show 01:42: Info vox voip controlplan 3 Outgoing call on voip id: 0, calling: 110, called: 111, call-id: 29. 01:42: Info vox voip controlplan 3 Alert in band received, call-id: 29. 01:42: Info vox voip userplan 3 VoIP RTP transmission started, coder: G729, call-id: 29. 01:42: Info vox voip userplan 3 VoIP RTP reception started, coder: G729, call-id: 29. 01:42: Info vox voip controlplan 3 Call connected, call-id: 29. 01:42: Info vox voip userplan 3 VoIP RTP transmission stopped, coder: G729, call-id: 29. 01:42: Info vox voip userplan 3 VoIP RTP reception stopped, coder: G729, call-id: 29. 01:42: Info vox voip userplan 3 Fax T38 starting call-id: 29 . 01:43: VOX.up.ifp NSF / CSI / DIS > 01:43: VOX.up.ifp.2 < TSI / DCS 01:43: VOX.up.ifp.2 < TCF (v29_9600) 01:43: Info vox voip userplan 1 T38 Pre-message procedure OK, call-id: 29. 01:43: VOX.up.ifp CFR > 01:43: Info vox voip userplan 1 T38 Transmitting page 1, call-id: 29. 01:43: VOX.up.ifp.2 < PAGE (v29_9600) 01:51: VOX.up.ifp.2 < PPS-EOP 01:51: Info vox voip userplan 1 T38 page 1 OK, call-id: 29. 01:51: VOX.up.ifp MCF > 01:51: VOX.up.ifp.2 < DCN 01:51: Info vox voip controlplan 3 Call Disconnection received on local port: 5/2, cause: (16)[Normal call clearing], call-id: 29. Copyright © OneAccess Networks – All rights reserved

Modem / fax pass-through event
Example of event in case of modem/fax call. One_training>event filter add vox all show 00:11: Info vox voip controlplan 3 Incoming call on voip id: 0, calling: 111, called: 110, call-id: 3. 00:11: Info vox voip controlplan 3 Outgoing call on local port: 5/0, calling: 111, called: 110, call-id:3. 00:11: Info vox voip controlplan 3 Alert received, call-id: 3. 00:11: Info vox voip controlplan 3 Call connected, call-id: 3. 00:11: Info vox voip userplan 1 Fax/Modem Passthrough starting call-id: 3. 00:11: Info vox voip userplan 3 RTP new transmission coder: G711 A Law, call-id: 3. 00:11: Info vox voip userplan 3 RTP new reception coder: G711 A Law, call-id: 3. 00:12: Info vox voip controlplan 3 Call Disconnection received on local port: 5/1, cause: (16)[Normal call clearing], call-id: 3. 00:12: Info vox voip userplan 3 VoIP RTP transmission stopped, coder: G711 A Law, call-id: 3. 00:12: Info vox voip userplan 3 VoIP RTP reception stopped, coder: G711 A Law, call-id: 3. Copyright © OneAccess Networks – All rights reserved

4.10 Voice Call Routing VRT (Voice Routing Table)

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4.10 Voice Call Routing VRT (Voice Routing Table)

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