Significance of $ in dial-peer destination pattern

$ means “end of digits” and is used for a more specific match. If you have two dial-peers with the same pattern and one with $ and the other without $, then call routing logic will always match the one with $, because that is more specific. Even if you give a lower preference to the dial-peer with $, still it will be matched because by default the dial-plan hunting algorithm is more specific match first and then preference.

dial-peer voice 10 pots

destination-pattern 1000

port 1/0/0

dial-peer voice 11 pots

destination-pattern 1000$

port 1/0/0

preference 5

In this case, always dial-peer 11 will be used to route calls to 1000, even though it has a lower preference. 

SIP messages between two endpoints.

Sep 11 00:41:59.171: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:1001@10.1.21.49:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.11.4:5060;branch=z9hG4bKBAEC
Remote-Party-ID: <sip:911@10.1.11.4>;party=calling;screen=no;privacy=off
From: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
To: <sip:1001@10.1.21.49>
Date: Mon, 11 Sep 2006 00:41:59 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
Supported: 100rel,timer,resource-priority
Min-SE:  1800
Cisco-Guid: 703035978-1080365531-2182204663-2878018674
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1157935319
Contact: <sip:911@10.1.11.4:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 238

v=0
o=CiscoSystemsSIP-GW-UserAgent 9779 8954 IN IP4 10.1.11.4
s=SIP Call
c=IN IP4 10.1.11.4
t=0 0
m=audio 19270 RTP/AVP 0 101
c=IN IP4 10.1.11.4
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Sep 11 00:41:59.179: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.11.4:5060;branch=z9hG4bKBAEC
From: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
To: <sip:1001@10.1.21.49>
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
CSeq: 101 INVITE
Server: Brekeke OnDO SIP Server (rev.172)
Content-Length: 0

Sep 11 00:41:59.215: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.11.4:5060;branch=z9hG4bKBAEC
From: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
To: <sip:1001@10.1.21.49>;tag=16777329
Date: Wed, 13 Sep 2006 12:47:08 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
Timestamp: 1157935319
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK
Allow-Events: telephone-event
Remote-Party-ID: <sip:1001@10.1.21.50>;party=called;screen=no;privacy=off
Contact: <sip:1001@10.1.21.49:5060>
Record-Route: <sip:10.1.21.49:5060;lr>
Content-Length: 0
Sep 11 00:42:00.559: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.11.4:5060;branch=z9hG4bKBAEC
From: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
To: <sip:1001@10.1.21.49>;tag=16777329
Date: Wed, 13 Sep 2006 12:47:08 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
Timestamp: 1157935319
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK
Allow-Events: telephone-event
Remote-Party-ID: <sip:1001@10.1.21.50>;party=called;screen=yes;privacy=off
Contact: <sip:1001@10.1.21.49:5060>
Record-Route: <sip:10.1.21.49:5060;lr>
Content-Type: application/sdp
Content-Length: 223

v=0
o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 10.1.21.50
s=SIP Call
c=IN IP4 10.1.21.50
t=0 0
m=audio 24678 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Sep 11 00:42:00.571: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:1001@10.1.21.49:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.11.4:5060;branch=z9hG4bKC1569
From: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
To: <sip:1001@10.1.21.49>;tag=16777329
Date: Mon, 11 Sep 2006 00:41:59 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
Route: <sip:10.1.21.49:5060;lr>
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0

Sep 11 00:42:03.447: %MV64340_ETHERNET-5-LATECOLLISION: FastEthernet0/1, late collision error
Sep 11 00:42:06.183: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:911@10.1.11.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.21.49:5060;rport;branch=z9hG4bKc06f1e40a2ac9bac.3
Via: SIP/2.0/UDP  10.1.21.50:5060;branch=z9hG4bK6134c268
From: <sip:1001@10.1.21.49>;tag=16777329
To: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
Date: Wed, 13 Sep 2006 12:47:09 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
User-Agent: Cisco-CCM4.1
Max-Forwards: 69
CSeq: 101 BYE
Record-Route: <sip:10.1.21.49:5060;lr>
Content-Length: 0

Sep 11 00:42:06.191: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.21.49:5060;rport;branch=z9hG4bKc06f1e40a2ac9bac.3,SIP/2.0/UDP  10.1.21.50:5060;branch=z9hG4bK6134c268
From: <sip:1001@10.1.21.49>;tag=16777329
To: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
Date: Mon, 11 Sep 2006 00:42:06 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 BYE
Reason: Q.850;cause=16
Content-Length: 0

How to debug DTMF relay?

For H323

debug h245 asn1

For SIP

debug voip rtp session named-event

Sample output

H323

Aug 21 21:07:37.902: h245_decode_one_pdu: more_pdus = 0, bytesLeftToDecode = 7
Aug 21 21:07:37.902: H245 MSC INCOMING ENCODE BUFFER::= 6D810447200063
Aug 21 21:07:37.902:
Aug 21 21:07:37.902: H245 MSC INCOMING PDU ::=

value MultimediaSystemControlMessage ::= indication : userInput : signal :
{
signalType “9”
duration 100
}

SIP

Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:04 00 00  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:04 00 00  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:04 00 00  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:04 01 90  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:84 03 20  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:84 03 20  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:84 03 20  <Snd>>>

How to prepend digits to caller id in H323?

voice translation-rule 1
rule 1 /\(^[2-9]………$\)/ /91\1/ — for national calls
rule 2 /\(.*\)/ /9011\1/ — for international calls

!
!
voice translation-profile ANI
translate calling 1

Creating a loopback dial-peer for testing

The IPIPGW supports configuration of an RTP loopback dial peer for use in verifying and troubleshooting H.323 networks. When a call encounters an RTP loopback dial peer, the gateway automatically signals call connect and loops all voice data back to the source. In contrast to normal calls through the VoIP-to-VoIP gateway, RTP loopback calls consist of only one call leg.

dial-peer voice 30 voip
destination-pattern 2222
session target loopback:rtp
incoming called-number 2222
codec g711ulaw

Telephony call-legs: 0
SIP call-legs: 0
H323 call-legs: 1
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 1

13FB : 67 97945490ms.1 +20 pid:30 Answer 5004 active
dur 00:00:12 tx:631/100960 rx:631/100960
IP 10.200.200.250:17012 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a

MGCP Call Preservation

MGCP PRI backhaul does not support call preservation when transitioned from Callmanager to SRST and vice versa.

H323/MGCP – Caller ID Display

H323 does not support Facility IE. H323 supports only Display IE

MGCP supports both Display and Facility IE

E1 R2 ANI

ANI is not available with E1R2 dtmf signaling. For all other E1R2 options, ANI is available and you need to select this option to have ANI send out.

H225 timeout for H323 dial peers

When setting up redundant h323 dial peers for sending calls from a H323 gateway to publisher and subscriber, set the h225 timeout to < 10 seconds. If its more than 10 seconds, q931 timer will expire and will do a call disconnect. If timeout is not set, the router waits 40 seconds to hunt between two dial-peers that have same pattern with  preference numbers in order.

 voice class h323 1

h225 timeout tcp establish 3 ——————— 3 seconds

dial-peer voice 1 voip

destination-pattern 3…

dtmf-relay h245-alphanumeric

pref 0 (default)

voice-class h323 1

session target ipv4:<ip of subscriber

dial-peer voice 2 voip

destination-pattern 3…

dtmf-relay h245-alphanumeric

pref 1

session target ipv4:<ip of publisher>

CFwdAll with Overlap Send receive

When OSR is enabled (for the 9. pattern), Cfwd all accepts only the digit 9 and call will be forwarded to 9. To avoid this configure a pattern 9.!, place that in a different partition and create a CSS with this partition in it.  Set the Cfwdall CSS on all ip phones to the newly created CSS.

How to enable Overlap Send-receive in CM 4.1

Overlap send receive is used in certain countries like Germany. Overlap send receive is supported in QSIG based switch types.

To setup OSR, add a route pattern 9. with Allow Overlap Sending enabled. Select the qsig gateway that you added (6608, mgcp t1 gateway etc). When OSR is enabled, FAC and CMC are automatically disabled.

Debug isdn q931 will look like this. The number dialed is 2142142142 The PSTN simulator router waits for each digit to come by and then routes the call. Make sure on the PSTN router the following command is enabled.

interface Serial1/0/0:23
no ip address
isdn switch-type primary-qsig
isdn overlap-receiving
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable

Jun 15 02:01:24.129: ISDN Se1/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x0001
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98384
Exclusive, Channel 4
Facility i = 0x9FAA06800100820100A11A020101020100801253796C7665737465722
05374616C6C6F6E65
Calling Party Number i = 0x0081, '5003'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '2' Plan:Unknown, Type:Unknown Jun 15 02:01:24.133: ISDN Se1/0/0:23 Q931: TX -> SETUP_ACK pd = 8 callref = 0x8
001
Channel ID i = 0xA98384
Exclusive, Channel 4
Jun 15 02:01:24.885: ISDN Se1/0/0:23 Q931: RX <- INFORMATION pd = 8 callref = 0
x0001
Called Party Number i = 0x80, '1'
Plan:Unknown, Type:Unknown
Jun 15 02:01:25.497: ISDN Se1/0/0:23 Q931: RX <- INFORMATION pd = 8 callref = 0
x0001
Called Party Number i = 0x80, '4'
Plan:Unknown, Type:Unknown
Jun 15 02:01:26.089: ISDN Se1/0/0:23 Q931: RX <- INFORMATION pd = 8 callref = 0
x0001
Called Party Number i = 0x80, '2'
Plan:Unknown, Type:Unknown
Jun 15 02:01:26.673: ISDN Se1/0/0:23 Q931: RX <- INFORMATION pd = 8 callref = 0
x0001
Called Party Number i = 0x80, '1'
Plan:Unknown, Type:Unknown
Jun 15 02:01:27.217: ISDN Se1/0/0:23 Q931: RX <- INFORMATION pd = 8 callref = 0
x0001
Called Party Number i = 0x80, '4'
Plan:Unknown, Type:Unknown
Jun 15 02:01:27.793: ISDN Se1/0/0:23 Q931: RX <- INFORMATION pd = 8 callref = 0
x0001
Called Party Number i = 0x80, '2'
Plan:Unknown, Type:Unknown
Jun 15 02:01:28.493: ISDN Se1/0/0:23 Q931: RX <- INFORMATION pd = 8 callref = 0
x0001
Called Party Number i = 0x80, '1'
Plan:Unknown, Type:Unknown
Jun 15 02:01:29.013: ISDN Se1/0/0:23 Q931: RX <- INFORMATION pd = 8 callref = 0
x0001
Called Party Number i = 0x80, '4'
Plan:Unknown, Type:Unknown
Jun 15 02:01:29.601: ISDN Se1/0/0:23 Q931: RX <- INFORMATION pd = 8 callref = 0
x0001
Called Party Number i = 0x80, '2'
Plan:Unknown, Type:Unknown
Jun 15 02:01:30.173: ISDN Se1/0/0:23 Q931: RX <- INFORMATION pd = 8 callref = 0
x0001
Called Party Number i = 0x80, '9'
Plan:Unknown, Type:Unknown
 

Translation rules and profiles

voice translation-rule 1

rule 1 /123/ /456/

rule 2 /^123/ /456/

rule 3 /^123$/ /456/

rule 4 /.*/ /456/

rule 5 /^123*/ /456/

rule 6 /^123+/ /456/

rule 7 /^123?/ /456/

rule 8 /^$/ /456/

a. rule 1 is a one to one replacement of any occurence of 123 in the source number with 456.

b. rule 2 replaces any number starting with 123 with a 456.

c. rule 3 replaces only the number 123 as the source number with 456.

d. rule 4 any number with the number 456, including null.

e. rule 5 says any number that starts with 12 and has 0 or more occurence of 3 with 456.

f. rule 6 says any number that starts with 12 and has 1 or more occurence of 3 with 456

g. rule 7 says any number that starts with 12 and has 0 or 1 occurence of 3 with 456.

h. rule 8 says any number with no input digits (empty ani for example) with 456.

Misc items:

A . (dot) means a single digit.

B. [0-9] specifies a range

C. .* means any digit followed by zero or more occurence, virtually any digit including null

D. .+ means any digit followed by one or more occurence, virtually any digit excluding null

E. ^$ means no digits.

How to block calls based on ANi

On a voice gateway create a translation rule as follows.

voice translation-rule 1

rule 1 reject /^$/

voice translation-profile CallBlock

translate incoming 1

dial-peer voice 1 pots

call-block translation-profile incoming CallBlock

Debug voice translation

Mar  1 09:09:40.338: //-1/xxxxxxxxxxxx/RXRULE/regxrule_get_profile_from_trunkgro
up_internal: Voice port 0x83564514 does not belong to any trunk group
Mar  1 09:09:40.342: //-1/519C1A918116/RXRULE/regxrule_match: Matched a call blo
ck rule; number= rule precedence=1
Mar  1 09:09:40.346: //-1/519C1A918116/RXRULE/regxrule_profile_block_internal: M
atched with rule 1 in ruleset 1
Mar  1 09:09:42.574: //-1/519C1A918116/RXRULE/regxrule_stack_pop_RegXruleNumInfo
: stack=0x8356487C; count=0

E1 R2 customizations

E1 R2 settings can be customized using a cas custom group.

controller e1 1/0

ds0-group 1 timeslots 1-24 type r2-signal r2-semi-compelled ani

cas-custom 1 —- this number should be same as ds0-group number.

dnis-digits min 3 max 11 —- if dialled digits maximum is not known, cas has to rely on a timeout (3 seconds by default). If max is known then cas will gain 3 seconds in inter digit timeout.

ani-digits min 1 max 65 —– if ani is blocked (callerid), this command can be used to block such calls. a minimum of 1 means there should be atleast one digit in the ani of the incoming call. After the router collects the max ani-digits, it sends the Caller ID End and does not collect any more digits for ani.

Refer to this link for additional custom parameters.

http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800942f2.shtml

Calling name in PRI

You can use either "Display IE" or "Facility IE" for Calling name.

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

FXO Caller ID

FXO Caller ID works only with H323 and not with MGCP

6608 MGCP configuration

To configure a 6608 blade to register with Callmanager, do the following,

a. Log into the 6500 switch and do a sh module to find the slot number in which 6608 is installed

b. Do a "set port voice interface help"

6500# (enable) set port voice interface help
Usage: set port voice interface #mod/port# dhcp enable [vlan #vlan#]
set port voice interface #mod/port# dhcp disable #ipaddrspec#
tftp #ipaddr# [vlan #vlan#]
[gateway #ipaddr#] [dns [ipaddr] [domain_name]]
(ipaddr_spec: #ipaddr# #mask#, or #ipaddr#/#mask#
#mask#: dotted format (255.255.255.0) or number of bits (0..31)
vlan: 1..4094
System DNS will be used if disabling DHCP without DNS parameters)

 For ex:

set port voice interface 7/2 dhcp disable 10.1.1.1/24 vlan #voicevlan# tftp 10.1.1.33 gateway 10.1.1.254 dns 10.1.1.2 cisco.com

or

set port voice interface 7/2 dhcp disable 10.1.1.1 255.255.255.0 vlan #voicevlan# tftp 10.1.1.33 gateway 10.1.1.254 dns 10.1.1.2 cisco.com

 c. Do a sh port <mod/port> and copy the mac-address of the port. Edit this in a notepad, and remove all
"-" (dashes).

 d. Use this mac-address to add a new gateway or add a new conf. bridge or transcoder.

MGCP messages

a. RSIP – Restart in Progress – When you reset a gateway this message is sent by gateway to CA.

b. AUEP and AUCX – Audit End point and Audit COnnection. – These messages are issued by the CA to the gateway. This is used to audit status of an endpoint and any connections associated with it.

c. CRCX, MDCX, DLCX – Create, Modify, Delete connection – Issued by CA to the gateway. CRCX is used to create a connection that terminates on the endpoint on the gateway. MDCX is used to modify parameters associated with the previously created connection. DLCX is used to delete an existing connection.

d. EPCF – End point configuration – Used by CA to send a configuration command to a gateway.

e. RQNT and NTFY – CA uses RQNT  (Request Notification) command to instruct the gateway to start monitoring for specific events such as hook actions or dtmf tones on a specified endpoint.

NTFY is used by gateway to inform the call-agent when the specified events take place.

 

MGCP UDP ports

MGCP uses UDP for sending clear text messages for control.

————Callmanager MGCP ports:——————–

CCM to gateway — uses src and destination UDP port 2427 (vice versa)

Gateway to CCM – Keepalive and backhaul port – TCP port 2428.

 

 

 

———- This is a general CA based on RFC 2705———- 

From gateways to CA uses UDP port 2727

From CA to gateways uses UDP port 2427

———————————————