IP Phone – Look and Feel across CM, CME and SRST

SRST to voicemail integration

In addition to supporting message buttons for retrieving personal messages, Cisco SRST allows the automatic forwarding of calls to busy and unanswered numbers to voice-mail systems. Voice-mail systems with BRI or PRI access can log in to the calling phone's mailbox directly. For this to happen, some Cisco CallManager configuration is recommended. If your voice-mail system supports Redirected Dialed Number Identification Service (RDNIS), RDNIS must be included in the outgoing SETUP message to Cisco CallManager to declare the last redirected number and the originally dialed number to and from configured devices and applications.

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Step 1 From any page in Cisco CallManager, click Device and Gateway.

Step 2 From the Find and List Gateways page, click Find.

Step 3 From the Find and List Gateways page, choose a device name.

Step 4 From the Gateway Configuration page, check Redirecting Number IE Delivery – Outgoing.

call-manager-fallback

voicemail 92145425600 —- when somebody hits message button it calls main site's voice mail pilot number

call-forward busy 92145425600 — for calls that are busy, redirect to main site's voice mail pilot number

call-forward noan 92145425600 timeout 15 — for calls that are not answered, redirect to main site's voice mail pilot number.

 

COR lists in SRST

call-manager-fallback

cor incoming <corlist> <list no> <list of numbers> ### applies corlist to outgoing calls from the IP phone to the router (SRST)

cor outgoing <corlist>  <list no> <list of numbers> ### applies corlist to incoming calls to the IP phones from router (SRST)

List number can be from 1-20. 

List of numbers should be separated by a dash and space to specify a range of phone extensions.

ex: cor incoming test 1 3001 – 3002

Applies cor list "test" and creates list 1 applied to numbers 3001 to 3002.

Default cor list

call-manager-fallback

cor incoming test default

This command applies cor-list Test to all incoming calls (w.r.t SRST router).

SRST Hunt groups

call-manager-fallback

no huntstop 

alias 1 9725425979 to 5001

alias 2 9725425979 to 5002

alias 3 9725425979 to 5003

pickup 9725425979

This way when a call comes to 9275425979, it will randomly route calls to one of the three numbers 5001,5002,5003. Using pickup softkey any phone can pick up the call ringning on the phones

Dial-peer hunting choices

dial-peer hunt ?
<0-7> Dial-peer hunting choices, listed in hunting order within each choice:
0 – Longest match in phone number, explicit preference, random selection. Default
1 – Longest match in phone number, explicit preference, least recent use.
2 – Explicit preference, longest match in phone number, random selection.
3 – Explicit preference, longest match in phone number, least recent use.
4 – Least recent use, longest match in phone number, explicit preference.
5 – Least recent use, explicit preference, longest match in phone number.
6 – Random selection.
7 – Least recent use.

SRST Alias

alias command also has a preference parameter. This is mostly used if you are entering specific matches rather than wild cards in alias command. For example, if you want to use an alias for IP Phone 4000 to send calls to 5000 if the phone is not registered, then make sure that you give a higher preference in alias as opposed to what is entered in the max-dn command

call-manager-fallback

max-dn 10 dual-line preference 1

alias 1 4000 to 5000 preference 5

Alias and Huntstop 

The alias huntstop keyword is relevant only if you have also set the global no huntstop command under call-manager-fallback. 

The alias huntstop keyword allows you to turn huntstop behavior back on for an individual alias, if huntstop is turned off globally by the no huntstop command. Setting the huntstop keyword on an individual alias stops hunting at the alias, making the alias the final member of the hunt sequence.

SRST call forwarding

SRST call-forwarding for busy and no answer can be set system wide

call-manager-fallback

call-forward busy <number>

call-forward noan <number> timeout 10

SRST vs CCM Callforwarding

Call-forward-all settings under regular Callmanager mode, is not retained by the phones when they go into SRST mode.

Call forward number transformations

You can also transform the extension number for call forwarding by using wild cards.

call-forward busy 40..

If a user calls 5010 and the user is busy, then the call will be forwarded to 4010.

Call forwarding on individual lines

alias 1 1001 to 1001 preference 1 cfw 2001 timeout 20

In this example, you have created a second dial peer for 1001 to route calls to 1001, but that has preference 1 and call forwarding to 2001. Because the preference on the dial peer created by the alias command is now a lower numeric value than the preference that the dial peer first created, all calls come initially to the dial peer created by the alias command. In that way they are subject to the forward as set by the alias command, instead of any call forwarding that may have been set globally. 

This feature can be used to create a basic hunt group setup using Callforwarding logic.

CME B-ACD

application

 service csq flash:app-b-acd-2.1.0.0.tcl

  param number-of-hunt-grps 3

  param aa-hunt2 2222

  param aa-hunt3 3333

  param aa-hunt4 4444

  param queue-len 15

  param queue-manager-debugs 1

!

 service aa flash:app-b-acd-aa-2.1.0.0.tcl

  paramspace english index 1

  paramspace english language en

  paramspace english location flash:

  param service-name csq

  param handoff-string aa

  param aa-pilot 6000

  param welcome-prompt en_bacd_welcome.au

  param number-of-hunt-grps 3

  param dial-by-extension-option 1

  param second-greeting-time 30

  param call-retry-timer 15

  param max-time-call-retry 120

  param max-time-vm-retry 2

  param voice-mail 4005

Define hunt groups 1,2,3 etc.

ephone-hunt 1 seq

pilot 1111

list 4001,4002

ephone-hunt 2 peer

pilot 2222

list 4003,4004

MGCP & H323 IOS gateway and T38 faxing

MGCP gateways and T38 

1. mgcp fax t38 ls_redundancy 0  OR

    mgcp fax t38 hs_redundancy 0

ls_redundancy set to 0 disables mgcp from sending redundant packets. The value can be set from  0  – 5 or 7 packets

hs_redundancy set to 0 disables mgcp from sending redundant packets for high speed fax m/c v.17, v27, v29 T.4 or T.6 fax machine. Default is 0, Can be configured up to 2 or 3.  0 disables redundancy.

Optional commands:

mgcp fax t38 gateway force  – This command forcest38 fax relay using NSE's eve if T38 and NSE's cannot be negotiated by the MGCP call agent at call setup time

mgcp fax t38 inhibit - Disables MGCP T38 fax relay.

mgcp tse payload <value>  - TSE is for Telephony signalling event. This command specifies payload value during fax and modem passthrough. Default is 100.

mgcp fax rate <rate>|voice  (Default rate is 7200 bps)

H323 gateways and T38

dial-peer voice 1 voip

fax rate voice

fax protocol t38 ls_redundancy 0

options available are : 

force - forces use of NSE;s to switch to T38 fax relay

fallback pass-through  g711ulaw | g711alaw - A fallback mode if T38 could not be negotiated.

MGCP IOS gateway and ATA 186 in H323 mode – Fax Passthrough configuration

1. Configure MGCP gateway as usual (with FXS ports).

2. Give the following commands

     no ccm-manager fax protocol   — disables cisco fax relay

     mgcp modem passthrough voip mode nse  — enables modem passthrough

     mgcp modem passthrough voip codec g711ulaw

MGCP default NSE payload types

Output of "Sh mgcp"

MGCP media bind ISABLED
MGCP Upspeed payload type for G711ulaw: 0,  G711alaw: 8 (You may opt to match this up with ATA NSE payload type)
MGCP Dynamic payload type for G.726-16K codec
MGCP Dynamic payload type for G.726-24K codec

3. Optionally you may want to disable T38 fax relay for MGCP (which is enabled by default)

mgcp fax t38 inhibit. (Useful if you want to just have passthrough enabled, and disable t38 fax on the gateway)

ATA 186 Fax modes (H323)

Fax pass through – Set audio mode to 0×00150015

Fax mode – (Use direct g711 only, it doesn't do any codec upspeed.) – Set audiomode to 0×00120012

12 – corresponds to 0001 0010

bit 1 – set to 1 for g711 only

bit 2 – set to 0 to disable CED

DSCP settings for H323 and MGCP gateways

The setting below applies to packets generated by teh router itself.

For H323 gateways

Default for signaling is af31 (26). Need to change this to cs3 (24).

dial-peer voice 1500 voip
ip qos dscp cs3 signaling 

For MGCP gateways

mgcp ip qos dscp cs3 signaling

Fax passthrough – H323 IOS gateway to ATA H323

** Please do not edit this blog due to table formatting issues **

Fax passthrough redundancy

Gateways will send each packet twice. Is useful if the network conditions are bad with packet loss and errors. Eradu packets send madithini.

ATA          Gateway         Results

Enabled     Enabled           Success

Enabled     Disabled          Success

Disabled    Enabled           Failure

Disabled    Disabled          Success

RTP Payload types

NTE and NSE are used to define a system of transporting tones and other telephony events over RTP, similar to DTMF relay

NTE                                         NSE

Named Telephony Event             Named Services Event

IETF – RFC 2833                       Cisco proprietary

RTP Payload 101                       RTP payload 100

Payload types for Fax upspeedThis is the payload type that tells the remote gateway to upspeed the codec to g711 for the fax call.

Default on ATA: 126/127 for fax upspeed to G.711m-law/G.711A-law

Default on Router: 0 for G711ulaw, 8 for G711alaw

You can change this in dial-peers

dial-peer voice 1500 voip

rtp payload-type cisco-pcm-switch-over-ulaw 126

rtp payload-type cisco-pcm-switch-over-alaw 127

ATA Settings

Redundancy

Connect mode: 0×90000480 (Bit 7 – 0/1 Disable/enable)

RTP Payload for upspeed to match IOS setting

Connect mode: 0×90000404 (Bit 2 – 0 means 126/127, 1 means 0/8)

Fax Passthrough over E1

ATA

ConnectMode bit 13 = 1 (g711alaw)

ConnectMode value = 0×90002400

IOS – H323
modem passthrough nse codec g711alaw

IOS – MGCP 

mgcp modem passthrough voip codec g711alaw 

VG248/6608

No settings, it will negotiate

IOS gateways – faxing – three steps

To enable fax passthrough in a H323 IOS gateway: (This can be given under a dial peer or globally under voice service voip mode)

Global mode:

1.  modem passthrough nse [payload-type number] codec {g711ulaw|g711alaw} [redundancy] [maximum-sessions <value>]

* Payload-type default : 100 for IOS gateways (Set this to same for all other gateways)

* Codec should be g711ulaw for fax sent over a t1 trunk or g711alaw for fax sent over a e1 trunk

* redundancy – enables fax redundancy. Set this parameter to be same at both ends

* max sessions – defines how many fax calls can occur simultaneously. Default is 16. Valid values 1 – 26.

2. Disable Fax relay – Fax relay is enabled by default and takes precedence over Passthrough.

fax rate disable

3. If you gave the modem passthrough setting under voice service voip mode, then you need to give the following command in dial-peer mode.

dial-peer voice 1 voip

modem passthrough system

Dial-peer mode: (2 steps)

 If you choose not to apply the modem passthrough settings in global mode, dial-peer voice 1 voip

fax rate disable

modem passthrough nse codec g711ulaw

To enable Fax passthrough in a Cisco IOS MGCP gateway

no ccm-manager fax protocol cisco

mgcp modem passthrough voip mode nse

mgcp modem passthrough voip codec g711ulaw|g711alaw.

H323 – ATA fax passthrough setting – recommendations

Audiomode - 0×00150015

0000 0000 0001 0101 (last 4 hex digits represents phone 1)

Bit 0 – 1 enables VAD

Bit 2 – 1 enables fax CED tone detection

Connect mode : 0×90000400

1001 0000 0000 0000 0000 0100 0000 0000

Bits 12 – 8 represent NSE payload type offset. Default NSE payload type is 100 for ATA which means the offset is set to 4 (00100). IOS H323/SIP gateways use NSE payload type 100. Most MGCP gateways using 6608 uses NSE payload type 101. So set the offset to 5 (00101) if setting up fax between ATA and 6608 (older versions)

Bits 13 – default 0 – represents fax pass through codec. 0 means g711ulaw, 1 means g711alaw.

To use fax in countries other than US, set

Connect mode : 0×90002400
**** key things to remember about interoperability of various AVVID fax components******

If a VG248, Cisco IOS router, MGCP gateway and ATA 186 is involved in faxing. Then set the NSE payload type to 100 (IOS Mode) in all devices. In VG248 set to IOS mode. In router by default NSE payload type is 100 (for modem passthrough). In MGCP IOS gateway (need to check). In 6608 set the mode to IOS mode. In ATA 186, default NSE payload type is 96 + offset of 4 = 100.

Fax passthrough – unlimited!!

What is Fax passthrough ?

Fax passthrough encodes fax traffic with in a g711 voice codec and sends it across the voip network as a voice call. The call may use any codec (g711, g729, g723) etc initially and once a 2100 Hz CED tone is detected, the device (ATA for ex:) tells the far end gateway to switch over to G711 using a peer-to-peer message. This message is called a NSE message (Named Signalling Event) with in the RTP stream.

ATA 186 H323 Audio parameters

LBRCodec:

If LBRCodec is set to 0, ATA chooses g723.1 as the Low bit rate codec. (You can have up to 2 calls on the ATA at g723.1)

If LBRCodec is set to 3, ATA chooses g729 as the Low bir rate codec (Only one call can be there on the ATA at g729). The allocation of g729 to either ports is dynamic. (Unlike SCCP, where the Connect mode bit 21 determines which port will use g729)

Audiomode 

Audiomode has 32 bits. First 16 bits set parameters for Phone port 1, Upper 16 bits set parameters for Phone port 2.

Default : 0×00150015

0000 0000 0001 0101 0000 0000 0001 0101

Bit 0 and 16 – Disable VAD (0 – Disable, 1 – Enable)

Bit 1 and 17 – if set to 1, enable g711 only. If set to 0 Enable LBR along with g711.

Bit 2 and 18 – if set to 1 enables fax CED tone detection

TxCodec:

To specify transmit audio codec preference

0 – g723 (only if LBRCodec is 0)

1 – g711alaw

2 – g711ulaw

3 – g729a (only if LBRCodec is 3)

RxCodec

0 – g723 (only if LBRCodec is 0)

1 – g711alaw

2 – g711ulaw

3 – g729a (only if LBRCodec is 3)

Calling between two ports on the same ATA

When in H323 mode, to place a call from Port 1 to port 2 the ATA should be configured as below.

1. Using gatekeeper

2. Using Callmanager as the gateway

3. Using an IOS gateway with h323-to-h323 hairpinning enabled 

CME – Gatekeeper registration

Ephone-dn's would register with the gatekeeper only if you are using the "dialplan-pattern" command in telephony-service.

telephony-service

dialplan-pattern 1 4085550100 extension-length 4 extension-pattern 4…

ephone-dn

number 4000 secondary 4001

Networking ATA, Callmanager and Voice gateway with FXS (phones)

Setup :

a. Callmanager 4.1 with IP phone extension 1005 (Callmanager IP 192.168.1.200)

b. ATA 186 with H323 load , 2 phones with extension 2000, 2001 (ATA IP 192.168.1.46)

c. H323 gateway with FXS ports with phone extension 3010, 3011 (Gateway IP 192.168.1.202)

d. Gatekeeper with 3 zones created (one for CM, one for gateway and one for ATA) (GK IP 192.168.1.201)

******Key things to note about GK registration*******

 a. ATA register with GK as Terminal

 b. H323 gateway registers with GK as Voip-GW

 c. Callmanager can be registered to the GK as Voip-GW or Terminal (Trunk configuration page)

 d. GK uses proxying between Terminals and Voip-GW's. So take special note to disable proxy from each zone that involes a Terminal and Voip-GW.

Configuration of ATA

a. Use E164 numbers in UID0 and UID1.

b. No need to set Login ID's if using E.164 numbers for GK registration

c. Specify the GKID field with the zone name (with out domain name)

d. Specify the GKOrProxy field with the IP address of the gatekeeper (192.168.1.201)

e. ATA cannot specify a tech prefix (as it registers as a terminal)

Configuration of GK

 

GateKeeper#sh gatek end
                    GATEKEEPER ENDPOINT REGISTRATION
                    ================================
CallSignalAddr  Port  RASSignalAddr   Port  Zone Name         Type    F
————— —– ————— —– ———         —-    –
192.168.1.46    1721  192.168.1.46    1739  ATA               TERM
    ENDPOINT ID: 8232D55C00000001  VERSION: 2 age= 233 secs
    E164-ID: 2001
192.168.1.46    1720  192.168.1.46    1719  ATA               TERM
    ENDPOINT ID: 82609CF000000002  VERSION: 2 age= 233 secs
    E164-ID: 2000
192.168.1.200   55419 192.168.1.200   53824 CM41              VOIP-GW
    ENDPOINT ID: 824FC0F400000004  VERSION: 2 age= 18 secs
    H323-ID: CCM_1
192.168.1.202   1720  192.168.1.202   50439 gateway           VOIP-GW
    ENDPOINT ID: 8230A09400000004  VERSION: 2 age= 40 secs
    H323-ID: 192.168.1.202
    E164-ID: 3011
    E164-ID: 3010
Total number of active registrations = 4

GateKeeper#

Gateway Configuration

interface Ethernet0/0
 ip address 192.168.1.202 255.255.255.0
 half-duplex
 h323-gateway voip interface
 h323-gateway voip id gateway ipaddr 192.168.1.201 1719
 h323-gateway voip h323-id Gateway
 h323-gateway voip tech-prefix 1#

dial-peer voice 1 pots
 destination-pattern 3010  —– phone 1
 port 1/0/0
!

dial-peer voice 2 pots
 destination-pattern 3011  —– phone 2
 port 1/0/1

dial-peer voice 10 voip
 destination-pattern 1…      —— Pattern for calls to Callmanager , No need to prepend tech prefix as 4# is  default tech prefix

 session target ras
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 
!
dial-peer voice 10 voip
 destination-pattern 2…     ——- Pattern for Calls to ATA, No need for tech prefix as ATA registers to GK as terminal
 session target ras
 dtmf-relay h245-alphanumeric
 codec g711ulaw
! 

gateway

Configuration of Callmanager

a. Add a gatekeeper (with GK's ip address, 192.168.1.201)

b. Add a Trunk (Gatekeeper controlled) and specify tech prefix 4#, the gatekeeper that was previously added, Terminal type (VOIP-GW or Terminal)  and zone name (CM41)

c. Set the significant digits on incoming calls to All.

d. Add a route pattern [23]XXX and route the calls through the trunk created in step 2. 2XXX is for ATA phones and 3XXX is for FXS phones

Basic H323 parameters – ATA in H323 mode

UID 0 – E.164 number of port 1, UID 1 - E.164 number of port 2. This is effective only if UseLoginID = 0.

LoginID 0 – H323 ID of port 1, LogingiD1 – H323 ID of port 2. This is effective only if UseLoginID = 1 (default 0)

Fax configuration

Configuring T 38 for MGCP gateway

mgcp fax t38 ecm

mgcp fax t38 gateway force

no mgcp fax t38 inhibit

Configuring T.38 for H323 gateway

dial-peer voice 10 voip

no modem passthrough

fax protocol t38 nse ls-redundancy 0 hs-redundancy 0 fallback cisco

 

Configuring Fax passthrough for H323 gateway

dial-peer voice 10 voip

modem passthrough nse codec g711ulaw

fax rate disable

Configuring Fax passthrough for MGCP gateway

mgcp modem passthrough voip mode nse

no ccm-manager fax protocol cisco

 

Configuring Fax relay for MGCP gateway

mgcp fax t38 inhibit

no mgcp fax t38 gateway force

ccm-manager fax protocol cisco

 

Configuring Fax relay for H323 gateway

dial-peer voice 10 voip

no modem passthrough

no fax rate disable

fax rate voice 

ATA 186 SCCP config page reference

If the pic is not legible, download the picture and zoom it to see more clearly.

How to reset ATA 186 to factory defaults

1. Pickup phone handset and press ATA function button

2. Dial FACTRESET#. The IVR will say press * to save or to exit press # key. Press * to save and it should reset the ATA to factory default configuration.

TOS bit setting (SCCP)

Default value :0×000068b8

 0110 1000 1011 1000 (ignoring leading 0's)

 0-7 digits represents RTP payload. Maps to DSCP 46 or TOS 5 (leading 3 digits 011)

8 – 15 digits represents Voice signalling. Maps to DSCP 26 or TOS 3 (leading 3 digits are 101)

To change the voice signalling to DSCP 24 (according to CM 4.1 specs) the new value of TOS is

0×000060b8     (Change bit 11)

Audiomode settings for ATA (SCCP)

Default value : 0×00350035

 32 bit field, first 16 bits for phone 1 and second 16 bits for phone 2 (starting from LSB)

0000 0000 0011 0101    0000 0000 0011 0101

 a. To force g711 on all calls set bit 1 and 17 to 1 – Audiomode becomes 0×00370037  (Both ports are forced to use g711)

 b. To use low bit rate codec or g711 (default setting) – set bits 1 and 17 to 0 – Audiomode becomes 0×003500035

Fax CED tone detection

Bits 2 and 18 should be set to 1 (default) to enable Fax CED tone detection

If set to 0, tone detection is disabled.

Codec Preference

If you use voice class codec to define preference of codecs, then the default codec of the dial peer will not be used in codec negotiation and you cannot define codecs directly on the dial-peer.
voice class codec 1

codec preference 1 g711ulaw

codec preference 1 g711alaw

In this case, the router will negotiate only g711ulaw and g711alaw. It will not use the dial-peer default codec of g729r8.

Fax between ATA on port 2 and Router with FXS (g729 codec)

Fax setup that works:

Fax on port 2 of ATA, LBRcodec = 3, Connect mode bit 21 = 1.

Router has dial-peer with g729r8 (br8) codec and fax machine on FXS port.

Commands in dial-peer:

dial-peer voice 10 voip

codec g729r8|br8

modem passthrough nse codec g711ulaw

fax rate disable

fax protocol pass-through g711ulaw

Fax setup that didnt work

Fax on port 2 of ATA, LBRcodec = 3, Connect mode bit 21 = 1.

Router has dial-peer with g729r8 (br8) codec and fax machine on FXS port.

Commands in dial-peer:

dial-peer voice 10 voip

codec g729r8|br8

fax rate disable

fax protocol pass-through g711ulaw

Conclusion : With out modem passthrough, Fax passthrough on routers are not working!

LBRCodec and Connect Mode (Skinny)

LBRCodec can take 2 values 0 or 3.

ATA can support following codec. g723, g729, g729a, g711(a/u).

Both ports can support g723 at same time

Both ports can support g711 at same time

Only one port can support g729 at any time.

Port 2 doesnt have support for g729a. Port 1 has.

When LBRCodec = 0,

port 1 and 2 can use g711ulaw, g711alawor g723.1

When LBRCodec =3 and Connect mode  bit 21 = 0 (affects only g729)

Port 1 can use g711ulaw, g711alaw, g729a or g729

When LBRCodec=3 and Connect mode bit 21 = 1 (affects only g729)

Port 2 can use g711ulaw, g711alaw, g729