voice-port 0/0/0
signal loopstart (or groundstart or E&M)
cptone <2DigitCountryCode> (set's call progress tones)
ring cadence pattern08 (or 'ring cadence define pulse interval)
ring frequency 50 (or 25 (Hz))
station_id number 3333 (caller ID)
station_id name Joe Smith (caller ID)
connection plar 4564 (auto ring-down to 4564)
impedance 600c (ohms) (or 600r (real), or 900c (complex), etc..)
input gain 2 (db) (scale back volume inbound)
output attenuation -1 (db) (scale back volume outbound)
echo-cancel coverage 32
!shut/no shut if you change signal type
voice-port 0/0/0
signal groundstart (or loopstart or E&M)
dial-type dtmf (or pulse)
connection plar opx 4564 (auto ring-down on off hook to 4564 - don't forget dial-peers to handle)
without the opx you could get stutter ring down
without the plar you would need to plan on 2-stage dialing or autoattendant plar
ring number <1-10> (number of rings before answering call (not nec with plar above))
!shut/no shut if you change signal type
voice-port 0/0/0
description FXS/DID
signal loopstart
signal did wink-start (or immediate-start, wink-start, delay-start)
!shut/no shut if you change signal type
dial-peer voice 12 pots
description inbound dialpeer
direct-inward-dial (alternate is plar or 2-stage dialing on FXO)
incoming called-number ....
port 0/0/0
dial-peer voice 13 pots
description outbound to phone for 3445 to port 0/0/1
destination-pattern 3445
port 0/0/1
voice-port 1/0/0
description fxo port for outbound dialing
signal groundstart
!shut/no shut if you change signal type
dial-peer voice 9 pots
description outbound to fxo port
destination-pattern 9[2-8]..........
forward-digits 10
port 1/0/0
voice-port 1/0/1
description CAMA 911 port to PSAP
signal cama KP-0-NPA-NXX-XXXX-ST (Type 2 CAMA Signaling)
KP-0-NXX-XXXX-ST (Type 1 CAMA Signaling)
KP-2-ST (Type 3 CAMA Signaling)
KP-II-NPA-NXX-XXXX-ST-KP-NPA-NXX-XXXX-ST
(Type 5 CAMA Signaling)
KP-NPD-NXX-XXXX-ST (Type 4 CAMA Signaling)
(may have to shut/no shut bounce port for CAMA signal config)
ani mapping 0 555 (for NPD, must config digit (here it's 0) for area code (here it's 555))
!shut/no shut if you change signal type
dial-peer voice 911 pots
destination-pattern 911
forward-digits all
port 1/0/1
dial-peer voice 9911 pots
destination-pattern 9911
forward-digits 3
port 1/0/1
voice-port 0/0/0
type 2 (or 1 or 3 or...See below for types)
operation two-wire (or four-wire)
signal immediate-start (or wink-start, or delay-dial)
!shut/no shut if you change signal type
term mon debug voice ccapi inout debug voip vtsp tone !was debug vtsp tone debug vpm signal gateway1#sh voice port sum IN OUT PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC =============== == ============ ===== ==== ======== ======== == 0/3/0 -- fxo-ls up dorm idle on-hook y 0/3/1 -- fxo-ls up dorm idle on-hook y you can look at port status and see if .ADMIN. state is .up. and .OPER. state is .dorm. and .OUT STATUS. is on-hook. This is a normal condition of the port. If OPER state shows .UP. then that means port is detecting off-hook/seizure or is stuck in disconnect mode. test voice port x/x/x si-reg-read 29 1 you will normally get either a 0X00 or 0XNN (where NN is some number other than zero). value of 0x00 indicates there is no voltage being seen on the line/voice port. You need the term mon to see the output of this command. test voice port 0/3/0 si-reg-read 29 1 gateway1# Apr 16 13:50:39.285: Values read from SiLabs Codec connected to DSP 0, channel 0: -------------------------------------------------------------- Register 29 = 0x00 simulating a call.................................... csim start <dn> gateway1# csim start 916175551212 csim: called number = 916175551212, loop count = 1 ping count = 0 csim err csimDisconnected recvd DISC cid(80) Apr 16 13:51:02.462: htsp_timer_stop3 htsp_setup_req Apr 16 13:51:02.466: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_HTSP_SETUP_REQ]fxols_onhook_setup Apr 16 13:51:02.466: [0/3/0] set signal state = 0xC timestamp = 0 Apr 16 13:51:02.466: htsp_timer - 1300 msec Apr 16 13:51:02.722: htsp_process_event: [0/3/0, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_1100]fxols_power_denial_detected Apr 16 13:51:02.722: htsp_timer2 - 1000 msec Apr 16 13:51:02.722: htsp_timer_stop csim: loop = 1, failed = 1 csim: call attempted = 1, setup failed = 1, tone failed = 0 gateway1# Apr 16 13:51:03.722: htsp_process_event: [0/3/0, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER2]fxols_power_den_disc Apr 16 13:51:03.722: htsp_timer_stop Apr 16 13:51:03.722: htsp_timer_stop2 Apr 16 13:51:03.722: [0/3/0] set signal state = 0x4 timestamp = 0 Apr 16 13:51:03.722: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_HTSP_RELEASE_REQ]fxols_onhook_release gateway1# sh call hist voice command will show a call attempt on that port.
controller T1 2/1
framing esf (or sf)
linecode b8zs (or ami)
clock source line (or internal)
ds-group 0 timeslots 1-12 e&m-wink-start (or e&m-immediate-start, or fxs-loop-start, etc...)
controller T1 2/1
framing esf (or sf)
linecode b8zs (or ami)
clock source line (or internal)
ds-group 1 timeslots 13 fxo-loop-start
voice-port 2/1:1
signal loopstart
voice-port 0/0/0
description fax
signal loopstart
connect fax1 voice-port 0/0/0 t1 2/1:1 (cross connect ds-group 1 to fxs port on 0/0/0)
dial-peer 211
incoming called number .
port 2/1:1
dial-peer 3000
destination-pattern 5554443000
port 0/0/0
show connection all (verify list of connections, mappings, current state)
isdn switch-type primary-dms100 (or primary-5ess, or primary-qsig, etc...)
controller T1 2/1
framing esf (or sf)
linecode b8zs (or ami)
clock source line (or internal)
pri-group timeslots 1-24 (could be fractional (e.g. between 1 to 5-15)
interface serial 2/1:23
isdn incoming-voice voice (set it to be processed by DSPs)
dial-peer voice 9 pots
destination-pattern 9[2-8].........
forward-digits 10
port 2/1:23
show voice port summary (show status on all voice ports)
network-clock-participate wic 4
network-clock-select 1 T1 0/3/0
card type t1 0 3
controller T1 0/3/0
cablelength long 0db
pri-group timeslots 1-24
interface Serial0/3/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn supp-service name calling
! send an alerting message before the connect message
isdn send-alerting
no cdp enable
no shutdown
!
voice-port 0/3/0:23
echo-cancel coverage 64
no shutdown
voice-port 1/1:0 echo-cancel enable (usually enabled by default) echo-cancel coverage 32 (ms, usually 64 by default)
voice vad-time 750 (ms - how long to wait before vad kicks in - 250ms in default) dial-peer voice 100 voip no vad (default enabled on dial-peers, but not on POTS interfaces) voice port 1/0 vad (default not enabled on POTS interfaces) comfort-noise (turn on white noise locally during VAD)
dial-peer voice 1 voip
incoming called-number .
dtmf-relay h245-alphanumeric
codec g711ulaw
dial-peer voice 6001 pots
destination-pattern 6001
port 0/0/0
dial-peer voice 6002 voip
destination-pattern 6002
session target ipv4:<ip addr> (could be ipv6: or dns:)
dial-peer voice 6003 voip
destination-pattern 6... (6 followed by 3 digits)
or
destination-pattern 6[3-5][3,5][3-5,7] (6, 2nd dig 3-5, 3rd dig 3 or 5, 4th dig 3-5 or 7)
or
destination-pattern 65(12)? (65 or 6512 will match)
or
destination-pattern 65(12)% (65 or 6512 or 651212 or 65121212 or ... up to 32 digits)
or
destination-pattern 65(12)+ (6512 or 651212 or 65121212 or ... up to 32 digits)
or
destination-pattern 9T (any dial pattern beginning with 9, up to 32 digits)
session target ipv4:<ip addr> (could be ipv6: or dns:)
dial-peer voice 6004 pots
destination-pattern 6...
(default strips named digits in pots)
no digit-strip
or
forward-digits 4
and/or
prefix 9,5554444 (comma pauses...)
port 0/0/0:23
num-exp 2... 4000 (change 2000-2999 to 4000 - direct it out 0/0/2 below) dial-peer voice 4000 pots destination-pattern 4000 port 0/0/2
voice translation-rule 1 rule 1 /3456/ /4444/ (replace 3456 with 4444) rule 2 /^\(...\)555\(....\)/ /\1444\2/ (replace nxx of 555 match with 444) rule 3 /^$/ /3000/ (replace anything (including NULL) with 3000) test voice translation-rule 1 6665553333
| ^ | start of string |
| $ | end of string |
| / | start or end of matching or replacement |
| \ | next char is special |
| [list] | list of chars |
| [^list] | (not) list of chars |
| . | single char |
| * | last reg exp 0+ times |
| + | last reg exp 1+ times |
| ? | last reg exp 0-1 time |
| () | group digits |
| & | all matched digits are to be added in replacement string |
voice translation-profile testtrunks-out translate called 1 (this could be calling or redirected-called) dial-peer voice 101 pots destination pattern 5.... port 0/0/0:23 translation-profile outgoing testtrunks-out voice-port 1/0:1 translation-profile incoming testtrunk-in
show dial-peer voice summary show dial-peer voice <#> show dialplan number 53322 (or any other number you want to test - show all settings that dialpeer will kick out) debug voip dialpeer debug voice translation
dial-peer voice 101 pots clid network-number 5554441212 second-number strip
| . | single digit 0-9 or * |
| [] | consecutive range [2-6] non-consecutive range [2,4,6] combination [2,4-6] |
| (match specific pattern) | |
| ? | preceding digit occurred 0 or 1 time |
| % | preceding digit occurfed 0 or more times |
| + | preceding digit occurred 1 or more times |
| T | wait a period of time to collect digits 0-9 or *. 15 seconds is default. # will end collection |
voice-card 1 codec complexity high (or medium - C549 choices)
voice-card 1 codec complexity flex (or high, medium, or secure - C5510/PVDM2 and PVDM3 choices)
show voice dsp (shows summary status of dsps installed
voice-card 1
dsp services dspfarm
dspfarm profile 10 transcode
codec g711ulaw
codec g729ar8
codec g729abr8
codec g729r8
maximum sessions 5
associate application SCCP
no shutdown
dspfarm profile 15 mtp (e.g. conferencing
codec g711ulaw
maximum sessions hardware 2
maximum sessions software 2 (e.g. use cpu)
associate application SCCP
no shutdown
sccp ccm 10.20.30.40 identifier 1 priority 1 version 7.0+
(priority - more than one can have diff priorities, version is CUCM version)
sccp local FastEthernet 3/0 (port that will communicate with CUCM?)
sccp ccm group 1
bind interface FastEthernet 3/0 (?)
associate ccm 1 priority 1
associate profile 15 (dspfarm profile above)
register TheDSPFARM1 (needs to match up on CUCM config)
CUCM - Media Resources/Conference Bridge menu
show voice port [<port num> | summary] (show details of analog voice port)
show controller [t1 | e1] (controller configuration)
show voice dsp (current dsp usage status)
test voice port [detector | inject-tone | loopback | relay | switch]
(test phys level pots port)
csim start <DN> (initiate outbound call, test dial-peers, translation rules, etc.)
debug dialpeer (review dialpeer matching in real time)
voice service voip
(shutdown [forced]) (shutdown service, forced means even if there are calls up)
h323
call start slow (this is global, default is fast)
session transport udp (default is tcp, not recommended to modify)
session transport tcp calls-per-connection 20 (1-9999, default is 15)
h225 timeout tcp call-idle value 5 (how long till tear down on idle call)
(0-1440 seconds, default is 10)
voice class h323 15 (set up a special class)
call start slow (slow start vs. fast start (default in later Cisco GWs))
h225 timeout tcp establish 3 (amount of time to hear response from remote gw - default is 15 seconds - use to speed up failover to backup gw)
h225 timeout tcp setup 3 (amount of time to hear response (to H.225 setup msg) from remote gw - default is 15 seconds - use to speed up failover to backup gw)
voice class codec 20
codec preference 1 g711ulaw bytes 160
codec preference 2 g711ulaw bytes 240
codec preference 3 g729br8
dial-peer voice 15 voip
voice-class h323 15 (use the elements in this voice class)
dial-peer voice 30
! remember initiating router will request perticular codec
! don't forget dial-peer 0 for inbound (all codecs supported)
voice-class codec 30 (can be used with H.323 MGCP, or SIP)
interface loopback0
ip address 10.11.12.13 255.255.255.0
h323-gateway voip bind srcaddr 10.11.12.13 (any outbound h.323 will use this address)
dial-peer voice 3000 voip destination pattern 3... session target ipv4:192.168.2.1
show gateway (what state and version is H.323 in) show h323 gateway h225 (show how many setup , alert, Progress, etc commands were sent received, and failed) clear h323 gateway h225 (clear the counters for the show command)
voice service voip
sip
session transport udp (or tcp)
bind control source-interface Loopback 0 (source-interface interface-id)
bind media source-interface Loopback 0
early-offer forced (force early offer every time, needs SW or HW MTP)
url sips (configure sip secure at global level)
exit
srtp (configure secure rtp at global level)
srtp fallback (configure fallback to rtp if srtp not supported)
CallerID - all can be in dial peer
signaling forward unconditional (forward display name with caller ID from PRI to terminating gw)
(make sure d-chan (23) interface has isdn supp-service name calling on it)
clid substitute name (clid number for name if none present)
no shutdown
sip-ua
authentication username <username> password <password> (digest auth)
registrar <name> expires <secs> (enable sip gw to reg e.164 nums on ext phones)
sip-server {dns:<hostname> | ipv4:<ipaddr>:[<portnum>]}
(with above you can specify session target sip-server in dial peer as opposed to interface addresses)
timers trying 1000 (wait for INVITE response for 1000ms)
timers connect 1000 (wait for ACK response for 1000ms)
timers expires 1000 (INVITE is valid for 1000ms)
retry {invite <number> | response <number> | bye <number> | cancel <number>}
bind all source-interface fa0/1 (can be control, media, or all - everything use this interface/ip addr)
dial-peer voice 40 voip
session protocol sipv2
destination pattern 3...
session target sip-server (defined in sip-ua section above)
voice-class sip url sips (configure sip secure for this dial peer - takes precedence over global)
dial-peer voice 90 voip
destination-pattern 9T
session target ipv4:<ipaddr>
session protocol sipv2
dtmf-relay {rtp-nte [digit-drop] | sip-notify} (out of band dtmf - good for low bw codecs)
clid strip pi-restrict (strip clid if it's designated as private on ISDN)
sip-ua
timers ?
more important | Range | Default | ||
| trying | time to wait for invite response | 100-1000ms | 500ms |
| connect | time to wait for ACK response | 100-1000ms | 500ms |
| disconnect | time to wait for BYE response | 60000-300000ms | 180000ms |
| expires | Time that an INVITE is valid | 100-1000ms | 500ms |
| Default | ||
| INVITE | max # of invite msg retries | 6 |
| RESPONSE | max # of response msg retries | 6 |
| BYE | max # of bye retries | 10 |
| CANCEL | max # of cancel retries | 10 |
voice service voip
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
registrar server expires max 600 min 60
!
dial-peer voice 1 voip
description test01 DID
preference 1
destination-pattern +180055512..
progress_ind setup enable 3
modem passthrough nse codec g711ulaw
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.2.10
incoming called-number .
dtmf-relay rtp-nte
no vad
!
dial-peer voice 2 voip
description Test02 DID
preference 2
destination-pattern +180055512..
progress_ind setup enable 3
modem passthrough nse codec g711ulaw
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.3.10
dtmf-relay rtp-nte
no vad
!
sip-ua
retry invite 2
retry response 2
retry bye 2
retry cancel 2
show sip service show sip-ua status (options, proxy, redirect, sdp settings) show sip-ua register status show sip-ua timers (default sip timers) show sip-ua retry (default sip retries) show sip-ua connections show sip-ua calls (SIP info on calls that are up) show sip-ua calls brief (lists all the call legs that are up) show call active voice summary show sip-ua statistics (SIP successes and failure counters)
mgcp package-capability ? as-package Select the Announcement Server Package atm-package Select the ATM Package dtmf-package Select the DTMF Package fm-package Select the FM Package fxr-package Select the FXR Package gm-package Select the Generic Media Package hs-package Select the Handset Package it-package Select the IT Package line-package Select the Line Package mdr-package Select the MDR Package mf-package Select the MF Package pre-package Select the PRE Package res-package Select the RES Package rtp-package Select the RTP Package script-package Select the Script Package srtp-package Select the SRTP Package sst-package Select the SST Package trunk-package Select the Trunk Package
mgcp ccm-manager mgcp mgcp call-agent 192.168.1.2 service-type mgcp or mgcp call-agent 192.168.1.2 2427 service-type mgcp version 0.1 mgcp package-capability line-package (default package for residential gws - clid, hook flash, reorder tones...) mgcp package-capability dtmf-package (dtmf tones) mgcp package-capability gm-package (generate media events, signal generic events such as congestion, fax tones, ringback) mgcp package-capability rtp-package (generage rtp event msgs such as continuity tones, tests, jitter buffer mod, RTP/RTCP timeouts
mgcp
ccm-manager mgcp
mgcp call-agent 192.168.1.2 service-type mgcp
or
mgcp call-agent 192.168.1.2 2427 service-type mgcp version 0.1
mgcp package-capability trunk-package
mgcp package-capability dtmf-package
mgcp package-capability gm-package
mgcp package-capability rtp-package
controller t1 1/0/1
ds0-group 0 timeslots 1-24 type none service mgcp
or
pri-group timeslots 1-24 service mgcp
show mgcp profile (basic config settings) show mgcp (status, timers, packages, codecs) show mgcp statistics (how many of each command successful and failed) show ccm-manager (make sure you're registered to CUCM)
debug voip rtp sess name show voip rtp connections
debug fax relay t30
class-map match-any media
match ip dscp ef
class-map match-any control
match ip dscp cs3
match ip dscp af31
class-map match-any qos-gateway-critical-traffic
match ip dscp cs6
!
!
policy-map voip
class media
bandwidth percent 50
class control
bandwidth percent 5
class qos-gateway-critical-traffic
bandwidth percent 5
class class-default
fair-queue
dial-peer voice 29 voip
destination-pattern [2-9].........
preference 1
session protocol sipv2
session target ipv4:<ipaddr>
voice-class codec 1
dtmf-relay rtp-nte sip-notify
ip qos dscp cs3 signaling
ip qos dscp ef media
voice-class sip options-keepalive
no vad
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
conf t
vlan 20
name Data
vlan 30
name Voice
int fa0/2 (or interface range fa0/5 - 12)
switchport mode access
switchport access vlan 20
switchport voice vlan 30
show vlan brief
small business SPA 300 phones SPA 500 phones - support SPCP (Smart Phone Control Protocol)- Cisc Proprietary - work with UC500 series platform - also suport SIP (e.g. offered by a ITSP (Internet Telephony Service Provider)) enterprise 9900 series 8900 series 7900 series 6900 series 3900 series - basic needs, public access areas
IP Communicator (Windows) Personal Communidcator - integrates voice voicemail, im, other features - WIndows and Mac Mobile Communicator - software package (integrates UC environment) on iPhone, Blackberry
7985G - hi res camera, color LCD 9951 - ehternet or wifi, touchscreen, hi res camera Video Advantage - application on Windows Cius
ATA 180 - 2 port adapters VG200 series adapters - VG224, 248 2, 4 - scheduled to replace
CER Unity/Unity ConnectoinCisco Converenc Connection suite billing applications
Call Manager/Unified Communications Manager UC Manager Business Edition - limit500 endpoints on eadchch appliance, no hgh availabilty/redundancy, integrated voicemail (unity connection) UC Manager Express - runs on Cisco routers - geared to business with up to 250 endpoints
| 2900 ISR | 3900 ISR | |
| max SRST calls | 250 | 1500 |
| max SIP sessions | 600 | 2500 |
| max dig voice galls | 400 | 660 |
| max FXO ports | 40 | 60 |
| max BRI ports | 24 | 38 |
100 ASR 9000 ASR 6500 7200 7600 12000 AS5400 AS5800
| E | Signaling output |
| M | Signaling output |
| SG | Signaling ground |
| SB | -48 volt signal battery |
| T | Audio input |
| R | Audio output |
| T1 | Secondary Audio input |
| R1 | Secondary Audio output |
| Type | Wires | Comments |
|---|---|---|
| I | 1 E, 2nd M, remaining 2 pairs audio PBX side - indicate off hook by connecting M to battery line side - indicate off hook by connecting E to ground |
most common in North America |
| II | 1 E, 2nd M, 3rd signal ground, 4th signal battery PBX side - indicate off hook by connecting M to SB (signal battery) Line side - indicate off hook by connecting E to SG (signal ground) |
used in sensitive environments - produces little interference |
| III | 4 wires for signaling idle - E open, M connect to SG PBX off hook - move M from SG to SB line side off hook - ground E |
not commonly used |
| IV | uses 4 wires for signaling idle - E and M open PBX off-hook, move M from SG to SB line side off-hook, move E to SG (grounded on PBX side) |
|
| V | similar to Type I. 2 wires (E & M) idle - both E&M are open. PBX off-hook - ground M line side - off hook - ground E |
most common outside of North America |
| SSDC5 | Similar to type V, but backwards - if line breaks, interface defaults to off-hook (busy) |
often found in England |
| H.225 Call Setup | Call Setup teardown, reformat Q931 to interoperate with H.225 msgs |
| H.225 Call Routing | Registration Administration Status (RAS) |
| H.235 | Security |
| H.245 | Logical transport channel. Capabilities exchange between endpoints |
| H.450 | Supplementary svcs - call divert, call hold, call park/pickup, call waitin |
| H.323 | ITU-T | P2P | Distributed |
| SIP | IETF | P2P | Distributed |
| MGCP | IETF | Client-Server | Centralized |
| SCCP | Cisco | Client-Server | Centralized |
conf t
voice service voip
fax protocol (can also go on dial-peer)conf t
dial-peer voice 1 voip
session protocol sipv2
fax protocol t38 ls-redundancy 3 hs-redundancy 3 fallback cisco (or pass-through)conf t
voice service voip
fax protocol pass-through g711ulaw
OR
fax protocol pass-through g711ulaw - on voip dial-peer
OR for mgcp
conf t
mgcp modem passthrough voip mode nse
mgcp modem passthrough voip codec g711alaw
mgcp package-capability rtp-package
mgcp fax t38 inhibit
conf t
dial-peer voice 1 voip
fax rate 9600 voice (setting max baud rate)
fax-relay ecm disable (disable error correction - typically on WAN links with dropped packets)
fax-relay ans-disable (disable SG3 answer tones - ends up dropping back to G3)
fax-relay sg3-to-g3 (negotiate G3 - same result as previous command)| 011 | International Access |
| 211 | Community government informatoin |
| 311 | City government informatoinIETF |
| 411 | Directory assistance |
| 511 | Traffic and road conditions |
| 611 | Telephone repair service |
| 711 | Hearing-disabled relay service |
| 811 | Underground pipe safety service |
| 911 | Emergency services |
voice-port 1/1:0 echo-cancel enable (usually enabled by default) echo-cancel coverage 32 (ms, usually 64 by default)
| PVDM Type | Low Complexity Call | Medium Complexity Calls | High Complexity Calls |
|---|---|---|---|
| PVDM (older C549) | 8 | 4 | 4 |
| PVDM2-8 (C5510) | 8 |
4 | 4 |
| PVDM2-16 | 16 | 8 | 6 |
| PVDM2-32 | 32 | 16 | 12 |
| PVDM2-48 | 48 | 24 | 18 |
| PVDM2-64 | 64 | 32 | 24 |
| PVDM Type | Low Complexity Call | Medium Complexity Calls | High Complexity Calls |
|---|---|---|---|
| PVDM3-16 | 16 | 12 | 10 |
| PVDM3-32 | 32 | 21 | 14 |
| PVDM3-64 | 64 | 42 | 28 |
| PVDM3-128 | 128 | 96 | 60 |
| PVDM3-192 | 192 | 138 | 88 |
| PVDM3-256 | 256 | 192 | 120 |
| Command | Meaning |
| AUEP | Audit endpoint |
| AUCX | Audit (endpoint) connection |
| CRCX | Create (RTP) conn that terms on GW |
| DLCX | Delete (RTP) conn that terms on GW |
| MDCX | modify (RTP) conn that terms on GW |
| RQNT | Request notification from VG for signaling events |
| EPCF | Endpoint config of VG by call agent |
| NTFY | Notify call agent of signaling event by VG |
| RSIP | Restart in progress - VG informs call agent that proc is restarting |
| Class | Usage | Min Power Level at Switch | Max power level at device |
|---|---|---|---|
| 0 | Default | 15.4 | 0.44-12.95 |
| 1 | Optional | 4.0 | 0.44-3.84 |
| 2 | Optional | 7.0 | 3.84-6.49 |
| 3 | Optional | 14.5 | 6.49-12.95 |
| 4 | Reserved for future use | N/A | N/A |
int s0/0/0:23
isdn negotiate-bchan (Enables the router to accept a B channel that is different from
the B channel requested in the outgoing call-setup message and
specifies the cause codes for which the call is reattempted.)
application
global
service alternate Default (specifies that default voice application takes over if
MGCP call agent is unavailable. Allows fallback to H.323
or SIP. Local dial peers will be considered for call routing).
ccm-manager fall-back-mgcp (enable srst for MGCP controlled ports).
voice translation-rule 2
rule 1 /^(4...\)/ /466\1/
voice translation-profile Internal
translate called 2
no telephony-service
call-manager-fallback (enter CM fallback cfg mode)
ip source-addr <ipaddr> (gw source addr)
max-ephones <num>
max-dn 12 dual-line
limit-dn <phonetype> <numDNs>
system message primary Help Me - Network is down
secondary-dialtone 9 (play dialtone after dialing 9)
translation-profile incoming internal
dial-peer voice 85101 pots
destination-pattern 851....
port 0/0/0:23
forward-digits all
prefix 401
show call-manager-fallback all
| Platform | SRST Number of Phones Supported | E-SRST Number of Phones Supported | Part Number (Spare) |
|---|---|---|---|
| Cisco 800 | Up to 4 phones | Up to 4 phones | - |
| Cisco 1861 | Up to 15 phones | Up to 15 phones | - |
| Cisco 2801 | Up to 25 phones | Up to 25 phones | FL-SRST-25= |
| Cisco 2811 | Up to 35 phones | Up to 35 phones | FL-SRST-35= |
| Cisco 2821 | Up to 50 phones | Up to 50 phones | FL-SRST-50= |
| Cisco 2851 | Up to 100 phones | Up to 100 phones | FL-SRST-100= |
| Cisco 3825 | Up to 350 phones* | Up to 175 phones | FL-SRST-175= |
| Cisco 3845 | Up to 730 phones** | Up to 250 phones | FL-SRST-250= |
| Cisco 2901 | Up to 35 phones | Up to 35 phones | FL-CME-SRST-5=, FL-CME-SRST-25=, FL-CME-SRST-100= |
| Cisco 2911 | Up to 50 phones | Up to 50 phones | FL-CME-SRST-5=, FL-CME-SRST-25=, FL-CME-SRST-100= |
| Cisco 2921 | Up to 100 phones | Up to 100 phones | FL-CME-SRST-5=, FL-CME-SRST-25=, FL-CME-SRST-100= |
| Cisco 2951 | Up to 250 phones | Up to 150 phones | FL-CME-SRST-5=, FL-CME-SRST-25=, FL-CME-SRST-100= |
| Cisco 3925 | Up to 730 phones | Up to 250 phones | FL-CME-SRST-5=, FL-CME-SRST-25=, FL-CME-SRST-100= |
| Cisco 3945 | Up to 1200 phones | Up to 350 phones | FL-CME-SRST-5=, FL-CME-SRST-25=, FL-CME-SRST-100= |
| Cisco 3925E | Up to 1350 phones | Up to 400 phones | FL-CME-SRST-5=, FL-CME-SRST-25=, FL-CME-SRST-100= |
| Cisco 3945E | Up to 1500 phones | Up to 450 phones | FL-CME-SRST-5=, FL-CME-SRST-25=, FL-CME-SRST-100= |
(config} # ip ssh version 2
(config} # ip ssh version 2
(config) # aaa new-model
(config) # aaa authentication login local_auth local
(config} # line vty 0 15
(config-line) # transport input ssh
(config-line) # login authentication local_auth