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The proxy server sendsa 100 Trying response immediately to the caller (Alice)
to stop the re-transmissions of the INVITE request.
The proxy server searches the address of Bob in the location server. After
getting the address, it forwards the INVITE request further.
A 200 OK response is generated soon after Bob picks the phone up.
Bob receives an ACK from the Alice, once it gets 200 OK.
At the same time, the session gets established and RTP packets (conversations)
start flowing from both ends.
After the conversation, any participant (Alice or Bob) can send a BYErequest to
terminate the session.
BYE reaches directly from Alice to Bob bypassing the proxy server.
Finally, Bob sends a 200 OK response to confirm the BYE and the session is
terminated.
The opening line of a request contains a method that defines the request, and a
Request-URI that defines where the request is to be sent.
Request Methods
SIP requests are the codes used to establish a communication. To
complement them, there are SIP responses that generally indicate
whether a request succeeded or failed.
These SIP requests which are known as METHODS make SIP message
workable.
METHODS can be regarded as SIP requests, since they request a specific action
to be taken by another user agent or server.
Core Methods
Extension Methods
Core Methods
There are six core methods as discussed below.
INVITE
INVITE is used to initiate a session with a user agent. In other words, an
INVITE method is used to establish a media session between the user
agents.
INVITE can contain the media information of the caller in the message body.
A successful INVITE request establishes a dialog between the two user agents
which continues until a BYE is sent to terminate the session.
INVITE Example
The following code shows how INVITE is used.
INVITE sips:Bob@TMC.com SIP/2.0
Via: SIP/2.0/TLS client.ANC.com:5061;branch = z9hG4bK74bf9
Max-Forwards: 70
BYE
BYE is the method used to terminate an established session. This is a SIP
request that can be sent by either the caller or the callee to end a session.
BYE request normally routes end to end, bypassing the proxy server.
REGISTER
REGISTER request performs the registration of a user agent. This request is
sent by a user agent to a registrar server.
It carries the AOR (Address of Record) in the To header of the user that is being
registered.
One user agent can send a REGISTER request on behalf of another user agent.
This is known as third-party registration. Here, the From tag contains the
URI of the party submitting the registration on behalf of the party identified in
the To header.
CANCEL
CANCEL is used to terminate a session which is not established. User agents
use this request to cancel a pending call attempt initiated earlier.
CANCEL is a hop by hop request, i.e., it goes through the elements between
the user agent and receives the response generated by the next stateful
element.
ACK
ACK is used to acknowledge the final responses to an INVITE method. An
ACK always goes in the direction of INVITE.ACK may contain SDP body
(media characteristics), if it is not available in INVITE.
ACK may not be used to modify the media description that has already been
sent in the initial INVITE.
A stateful proxy receiving an ACK must determine whether or not the ACK
should be forwarded downstream to another proxy or user agent.
For 2xx responses, ACK is end to end, but for all other final responses, it works
on hop by hop basis when stateful proxies are involved.
OPTIONS
OPTIONS method is used to query a user agent or a proxy server about its
capabilities and discover its current availability. The response to a request
lists the capabilities of the user agent or server. A proxy never generates an
OPTIONS request.
Extension Methods
Subscribe
SUBSCRIBE is used by user agents to establish a subscription for the
purpose of getting notification about a particular event.
After the time period passes, the subscription will automatically terminate.
You can re-subscription again by sending another SUBSCRIBE within the dialog
before the expiration time.
NOTIFY
NOTIFY is used by user agents to get the occurrence of a particular event.
Usually a NOTIFY will trigger within a dialog when a subscription exists
between the subscriber and the notifier.
NOTIFY
contain
an Event header
field
indicating
the
event
and
PUBLISH
PUBLISH is used by a user agent to send event state information to a
server.
PUBLISH is mostly useful when there are multiple sources of event information.
PUBLISH
request
must
contain
an Expires header
field
and
a Min-
REFER
REFER is used by a user agent to refer another user agent to access a URI
for the dialog.
REFER must contain a Refer-To header. This is a mandatory header for REFER.
A 202 Accepted will trigger a REFER request which indicates that other user
agent has accepted the reference.
INFO
INFO is used by a user agent to send call signalling information to another
user agent with which it has established a media session.
UPDATE
UPDATE is used to modify the state of a session if a session is not
established. User could change the codec with UPDATE.
PRACK
PRACK is used to acknowledge the receipt of a reliable transfer of
provisional response (1XX).
The PRACK method applies to all provisional responses except the 100 Trying
response, which is never reliably transported.
MESSAGE
It is used to send an instant message using SIP. An IM usually consists of
short messages exchanged in real time by participants engaged in text
conversation.
The
contents
of
MESSAGE
are
carried
in
the
message
body
as
a MIMEattachment.
A 200 OK response is normally received to indicate that the message has been
delivered at its destination.
text, these texts can be gibberish unless you understand fully what they
mean.
This document attempts to break down each component of the SIP
interaction using a practical approach. We will look at various logs, the SIP
messages, headers, SDP information and try to figure out what is going on in
a sip voice call transaction.
In as much as I will try to define the under lying layer of the SIP messaging,
this document will not go into in-depth analysis of the SIP protocol, so it is
advisable to understand SIP protocol technology to be able to understand sip
traces.
One key element of troubleshooting is this: To fix a problem, you need to
understand the issue, how it works before you can restore it to order.
One popular debug used in troubleshooting a sip solution on a cisco IOS
router is
Call-ID
CSeq
Contact
Max-Forwards
Expires
via.png
In SIP responses follow the via header except for future requests like ACK and BYE where
responses are sent to the contact header
A key component of the sip message. Its tells us the time of the sip request.
Call ID:
Call-ID: 68781700-f791ec0f-2d26-e28690a@10.105.80.114
The Call-ID header field is an identifier used to keep track of a particular SIP
session. The originator of the request creates a locally unique string. Some
older implementations also add an @ and its host name to the string. The
initiator of the session that generates the establishing INVITE generates the
unique Call-ID and From tag.
Cseq Header:
CSeq: 101 INVITE
The command sequence CSeq header field is a required header field in every
request. The CSeq header field contains a decimal number that increases for
each request. Usually, it increases by 1 for each new request, with the
exception of CANCEL and ACK requests, which use the CSeq number of the
INVITE request to which it refers. The CSeq count is used by UASs to
determine out-of-sequence requests or to differentiate between a new
request (different CSeq) or a retransmission (same CSeq). The CSeq header fi
eld is used by UACs to match a response to the request it references
User-Agent: Cisco-CUCM8.6
specifies that the attributes containing a=rtpmap should be used for each
media field
Version Number
Origin
s=SIP Call
c=IN IP4 10.133.92.102
Call Subject
Connection/IP address for RTP
stream
time
Media
Attributes-media
Attributes-Packetization
Dtmf attributes
t=0 0
m=audio 25268 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephoneevent/8000
a=fmtp:101 0-15
Dtmf tones
This line defines the media attribtes that will be used for the call.
Audio:
means that this is an Audio call, we can also have
m=video in case of a Video call
25268:
RTP/AVP:
Represents the RTP/AVP profile number for each of the
profiles listed. The profile numbers are explained below
18=G729
0=PCMU
8=PCMA
101=rtp-nte payload
Now we have looked at the basics of sip headers and messages, lets use this
to understand the following sip trace
The call flow for this call is as shown:
PSTN-------->ITSP------->CUBE--------------->CUCM---------------->IP PHONE
ITSP: 10.10.33.132
CUBE:10.100.0.74
CUCM:10.100.0.14
1. An inbound call is received on the CUBE from the ITSP. This invite was sent
with SDP. NB that this inbound leg of this call will have a unique call ID that
shows the origin of the call, highlighted below.
Received:
INVITE sip:441127653485@10.100.0.74:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.33.24:5070;branch=z9hG4bK9377fo00cg5ha7l0g3t0.1
From: <sip:07455900064@212.136.178.216:5060;user=phone>;tag=1526438727-
1338998848384To: "voice-lab-aokanlawon"<sip:441127653485@pbx.emea.ipcom.com>
Call-ID: BW1807283840606121067600210@212.136.178.216
CSeq: 558267841 INVITE
Contact: <sip:07455900064@10.10.33.24:5070;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Accept: multipart/mixed,application/media_control+xml,application/sdp
Supported:
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 207
v=0
o=BroadWorks 161384582 1 IN IP4 10.10.33.132
s=c=IN IP4 10.10.33.132
t=0 0
m=audio 11164 RTP/AVP 18 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=fmtp:18 annexb=no
+++From our understanding of the traces, we see that the call originates
from a device called Broadworks, which advertises G711a, G711u, G729 and
uses rtp-nte for DTMF. We also see the IP address for the CUBE to stream its
RTP to.+++++
3. Next the CUBE sends a trying to ITSP. Trying simply means: I am looking
for the number you have requested.
4. Next the CUBE receives a trying from CUCM. The call-ID help us to know
where these responses are coming from.
002793: Jun 6 16:07:28.396: //260863/8C38FA5E95E3/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.100.0.74:5060;branch=z9hG4bK7953C1859
From: <sip:07455900064@10.100.0.74>;tag=4C85762C-1A2D
To: <sip:901127653485@10.100.0.14>
Date: Wed, 06 Jun 2012 16:07:28 GMT
Call-ID: 8C394872-AF2811E1-95E98F4D-5D7E5E41@10.100.0.74
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0
5. Next the CUBE receives ringing from CUCM This informs the CUBE that the
called endpoint is ringing
002794: Jun 6 16:07:28.412: //260863/8C38FA5E95E3/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.100.0.74:5060;branch=z9hG4bK7953C1859
From: <sip:07455900064@10.100.0.74>;tag=4C85762C-1A2D
To: <sip:901127653485@10.100.0.14>;tag=811674~ffa80926-5fac-4dd6-b405-2dbbc56ae9a2477917854
Date: Wed, 06 Jun 2012 16:07:28 GMT
Call-ID:
8C394872-AF2811E1-95E98F4D-5D7E5E41@10.100.0.74
7. Now the CUBE receives a 200 ok from CUCM. Please note that some
elements of the SDP has changed
c=IN IP4 10.100.20.10------------------------IP address to send RTP stream to
t=0 0 -------------------------------------------Duration of the call
m=audio 16730 RTP/AVP 18 101---------Codec to use for call and DTMF type
to use
a=rtpmap:18 G729/8000-------------------Codec = G729
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
9. CUBE then sends a 200 OK to ITSP with the SDP attributes to use for the
Call based on what it received from CUCM.
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.100.0.14:5060;branch=z9hG4bK198a0b7ee5d33c
From: <sip:901127653485@10.100.0.14>;tag=811674~ffa80926-5fac-4dd6-b4052dbbc56ae9a2-477917854
To: <sip:07455900064@10.100.0.74>;tag=4C85762C-1A2D
Date: Wed, 06 Jun 2012 16:07:28 GMT
Call-ID: 8C394872-AF2811E1-95E98F4D-5D7E5E41@10.100.0.74
CSeq: 101 SUBSCRIBE
Content-Length: 0
Contact: <sip:07455900064@10.100.0.74:5060;transport=tcp>
Expires: 7200
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.33.24:5070;branch=z9hG4bK9377fo00cg5ha7l0g3t0.1
From: <sip:07455900064@212.136.178.216:5060;user=phone>;tag=15264387271338998848384To: "voice-lab-aokanlawon"<sip:441127653485@pbx.emea.ipcom.com>;tag=4C85763E-1CF8
Call-ID:
BW1807283840606121067600210@212.136.178.216
11. Finally the call is ended. Now when troubleshooting the direction of call
termination is important. In this case we can see that the CUBE receives a
BYE, which is the sip method for call termination. However who sent the BYE,
is it CUCM or ITSPThe answer is in the Call-ID. As we call can see the CALLID is for the leg from the ITSP. So we see that the call was terminated from
the ITSP side.
Next important thing is the cause code. The reason why the call was
terminated.
CSeq: 558267842 BYE
Reason: Q.850;cause=16
Here we see as normal call clearing Cause=16.
Received:
BYE sip:441127653485@10.100.0.74:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.33.24:5070;branch=z9hG4bKfj8rji3008r0m4lbg7e0cd1hhq713.1
From: <sip:07455900064@212.136.178.216:5060;user=phone>;tag=15264387271338998848384To: "voice-lab-aokanlawon"<sip:441127653485@pbx.emea.ipcom.com>;tag=4C85763E-1CF8
Call-ID: BW1807283840606121067600210@212.136.178.216
CSeq: 558267842 BYE
Max-Forwards: 69
Content-Length: 0
Received:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.100.0.74:5060;branch=z9hG4bK7954021A8
From: <sip:07455900064@10.100.0.74>;tag=4C85762C-1A2D
To: <sip:901127653485@10.100.0.14>;tag=811674~ffa80926-5fac-4dd6-b405-2dbbc56ae9a2477917854
Date: Wed, 06 Jun 2012 16:07:34 GMT
Call-ID: 8C394872-AF2811E1-95E98F4D-5D7E5E41@10.100.0.74
CSeq: 103 BYE
Content-Length: 0
Below are a few of the Threads that we have used the indepth understanding
of sip trcaes to help resolve thier issues. Please take a look as this will help
you to understand better sip traces and how they play a key part in
troubleshooitng issues
https://supportforums.cisco.com/message/3646952#3646952
https://supportforums.cisco.com/message/3634572#3634572
https://supportforums.cisco.com/message/3624258#3624258
https://supportforums.cisco.com/message/3653801#3653801
Support
...
Troubleshoot and Alerts
Configuration Example and TechNotes
Understanding H.323
Gatekeepers
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View with Adobe Reader on a variety of devices
Updated:Sep 25, 2014
Document ID:5244
Contents
Introduction
Prerequisites
Requirements
Components Used
Conventions
Gatekeeper Definition
Gatekeeper Zones and Subnets
Gatekeeper Functionality
Mandatory Gatekeeper Functions
Optional Gatekeeper Functions
H.323 Protocol Suite
H.225 RAS Signaling
H.225 Call Control (Setup) Signaling
H.245 Media Control and Transport
H.323 Protocol Suite Overview
H.225 RAS Signaling: Gatekeepers and Gateways
RAS Gatekeeper Discovery
RAS Registration and Unregistration
RAS Admissions
RAS Endpoint Location
RAS Status Information
RAS Bandwidth Control
Gatekeeper-Routed Call Signaling vs Direct Endpoint Signaling
Introduction
The ITU-T H.323 standard specifies four components:
gateway
gatekeeper
terminal
Prerequisites
Requirements
Ensure that you use the H.323 Gatekeeper functionality feature, which is denoted as x- on
the Downloads(registered customers only) . For example, a valid Cisco IOS for the Cisco
2600 to act as a gatekeeper is c2600-ix-mz.122-11.
Components Used
This document is not restricted to specific software and hardware versions.
Conventions
Refer to the Cisco Technical Tips Conventions for more information on document
conventions.
Gatekeeper Definition
A gatekeeper is an H.323 entity on the network that provides services such as address
translation and network access control for H.323 terminals, gateways, and MCUs. Also, they
can provide other services such as bandwidth management, accounting, and dial plans that
you can centralize in order to provide salability.
Gatekeepers are logically separated from H.323 endpoints such as terminals and gateways.
They are optional in an H.323 network. But if a gatekeeper is present, endpoints must use
the services provided.
Gatekeeper Functionality
The H.323 standard defines mandatory and optional gatekeeper functions:
Call AuthorizationWith this option, the gatekeeper can restrict access to certain
terminals or gateways and/or have time-of-day policies restrict access.
Call Control SignalingWith this option, the gatekeeper can route call-signaling
messages between H.323 endpoints with the use of the Gatekeeper-Routed Call Signaling
(GKRCS) model. Alternatively, it allows endpoints to send H.225 call-signaling messages
directly to each other.
Note: Cisco IOS gatekeepers are Direct Endpoint Signaling based. They do not support
GKRCS. See theGatekeeper-Routed Call Signaling vs Direct Endpoint Signaling section of this
document.
RAS uses User Datagram Protocol (UDP) ports 1719 (H.225 RAS messages) and 1718
(multicast gatekeeper discovery).
See the H.225 RAS Signaling: Gatekeepers and Gateways section of this document for more
detailed information.
Signaling vs Direct Endpoint Signaling section of this document for more information. The
method chosen is decided by the gatekeeper during the RAS admission message exchange.
If no gatekeeper is present, H.225 messages are exchanged directly between the endpoints.
flow control
If an H.323 endpoint does not know its gatekeeper, then it can send a Gatekeeper
Request (GRQ). This is a UDP datagram addressed to the well-known destination port 1718
and transmitted in the form of an IP multicast with the multicast group address
224.0.1.41.
One or several gatekeepers can answer the request with either a positive Gatekeeper
Confirmation (GCF) message or a negative Gatekeeper Reject (GRJ) message. A reject
message contains the reason for the rejection and can optionally return information about
alternative gatekeepers. Auto discovery enables an endpoint to discover its gatekeeper
through a multicast Gatekeeper Request (GRQ) message. Because endpoints do not have
to be statically configured for gatekeepers, this method has less administrative overhead.
A gatekeeper replies with a GCF or GRJ message. A gatekeeper can be configured to
respond only to certain subnets.
Note: A Cisco IOS gatekeeper always replies to a GRQ with a GCF/GRJ message. It never
remains silent.
If a gatekeeper is not available, the gateway periodically attempts to rediscover a
gatekeeper. If a gateway discovers the gatekeeper has gone off-line, it ceases to accept new
calls and attempts to rediscover a gatekeeper. Active calls are not affected.
Gatekeeper Discovery
GRQ (Gatekeeper_Request)
GCF (Gatekeeper_Confirm)
A message sent
by endpoint to
gatekeeper.
A reply from
gatekeeper to
endpoint which
indicates the
transport address
of the gatekeeper
RAS channel.
GRJ (Gatekeeper_Reject)
A reply from
gatekeeper to
endpoint that
rejects the
endpoint's
request for
registration.
Usually due to
gateway or
gatekeeper
configuration
error.
This table defines the RAS gatekeeper registration and unregistration messages:
Gatekeeper Discovery
RRQ (Registration_Request)
Sent from an
endpoint to a
gatekeeper RAS
channel address.
RCF (Registration_Confirm)
RRJ (Registration_Reject)
URQ (Unregister_Request)
Sent from
endpoint or
gatekeeper to
cancel
registration.
UCF (Unregister_Confirm)
Sent from
endpoint or
gatekeeper to
confirm an
unregistration.
URJ (Unregister_Reject)
RAS Admissions
Admission messages between endpoints and gatekeepers provide the basis for call
admissions and bandwidth control. Gatekeepers authorize access to H.323 networks with the
confirmation of or rejection of an admission request.
This table defines the RAS admission messages:
Admission Messages
ARQ (Admission_Request)
An attempt by an
endpoint to
initiate a call.
ACF (Admission_Confirm)
An authorization
by the
gatekeeper to
admit the call.
This message
contains the IP
address of the
terminating
gateway or
gatekeeper and
enables the
original gateway
to initiate call
control signaling
procedures.
ARJ (Admission_Reject)
Denies the
request of the
endpoint to gain
access to the
network for this
particular call.
See the Gatekeeper to Gateways Call Flow section of this document for more information.
Location Request
LRQ (Location_Request)
LCF (Location_Confirm)
Sent to request
the gatekeeper
contact
information for
one or more
E.164 addresses.
Sent by the
gatekeeper and
contains the call
signaling channel
or RAS channel
address of itself
or the requested
endpoint. LCF
uses its own
address when
GKRCS is used.
LCF uses the
requested
endpoint address
when Directed
Endpoint Call
Signaling is used.
LRJ (Location_Reject)
Sent by
gatekeepers that
received an LRQ
for which the
requested
endpoint is not
registered or has
unavailable
resources.
See the Gatekeeper to Gateways Call Flow section for more information.
Status Information
IRQ (Information_Request)
IRR
(Information_Request_Response)
A status
request
sent from
gatekeepe
r to
endpoint.
Sent from
endpoint
to
gatekeepe
r in
response
to IRQ.
This
message is
also sent
from
endpoint
to
gatekeepe
r if the
gatekeepe
r requests
periodic
status
updates.
The IRR is
used by
gateways
to inform
the
gatekeepe
r about the
active
calls.
IACK (Info_Request_Acknowledge)
Used by
the
gatekeepe
r in order
to respond
to IRR
messages.
INACK
(Info_Request_Neg_Acknowledge)
Used by
the
gatekeepe
r in order
to respond
to IRR
messages.
Bandwidth Control
BRQ (Bandwidth_Request)
A request for an
increase/decrease in
call bandwidth sent
by the endpoint to
the gatekeeper.
BCF (Bandwidth_Confirm)
Sent by the
gatekeeper and
confirms the
acceptance of the
bandwidth change
request.
BRJ (Bandwith_Reject)
Sent by the
gatekeeper and
rejects the
bandwidth change
request.
This is used by
gateways to inform
the gatekeeper
whether resources
are available in the
gateway to take on
additional calls.
Call Disconnect
This diagram illustrates the concept of VoIP Network scaling with gatekeepers and directory
gatekeepers:
Note: Refer to Understanding Cisco IOS Gatekeeper Call Routing for more information on
gatekeeper sample configurations.
Configure presence groups with which watchers can be associated, that specify rules for viewing the
presence status of presence entities that are associated with another group.
The first aspect of presence policies for CUCM is the subscribe CSS. CUCM uses the subscribe CSS to determine
how to route presence requests. Presence requests are SUBSCRIBE messages with the Event field set to Presence.
These messages are sent from the watcher, which can be a phone or a trunk. The subscribe CSS is associated with
the watcher and lists the partitions that the watcher is allowed to see. This mechanism provides an additional level of
granularity for the presence SUBSCRIBE requests to be routed independently from the normal call-processing CSS.
With the subscribe CSS set to <None>, BLF speed dial and call list presence status does not work (if no directory
number or route pattern is associated with the <None> partition) and the subscription message is rejected as user
unknown. When a valid subscribe CSS is specified, the indicators work and the SUBSCRIBE messages are
accepted and routed properly.
Translation patterns are used for modifying the calling/called party number based on the
dialed number. This is done before a destination is selected, meaning calls route through
translation patterns to modify the calling/called number before reaching a route pattern which
points to a gateway or an end device such as a phone. Transformation patterns are modifying
the calling/called party number once an endpoint or destination has been selected.
Transformation patterns are used to change the calling/called party number right before
a call is sent out of a gateway or trunk, or can be used to change the displayed number on a
phone when a call is received.
Translation used for digit manipulation before (pre-routing decision made) a destination(GW,
Trunk ) is chosen.
Transformation used for digit manipulation after routing decision made like GW trunk is chosen.
so its pre-routing for translation and post-routing for transformation which i can say.
Adding to my previous response in simple words Translation Pattern has the ability to change
the destination of a call whereas Transformation Pattern can only change the display i.e., how
Calling and Called number will show up as on the phone.
1. Translation Patter applies on DNIS - You mean to say incoming call to CUCM only or is it
managed with CSS for outbound as well.
2. Translation is kind of a rule where transformation is kind of a change on a number mask. Am I
correct? Correct me If I am wrong.
Below is a example which I would like to understand.
1. You can apply TP to both inbound and outbound calls. Moreover to internal calls as well.
Beauty of TP is it doens't select the trunk/gateway and you always have a choice where to take
this call further. And in either case, whether to apply or not can be controlled through CSS.
2. Not exactly. If you see the manipulation parameters under TP (with respect to called party
number) and Called Party Transformation, all manipulation parameters are same. Difference is
at which part of call flow, you apply the same.
In your example, let see how we can apply either translation pattern or called party
transformation to strip prefix viz 91.
1. If we use translation pattern, instead of creating RP like 91.[2-9]xxxxxxxxx, create TP as 91.
[2-9]xxxxxxxxx. Now under translation pattern, strip predot. This will change the number to [29]xxxxxxxxx
Since TP can't decide the gateway/trunk, you need a RP like [2-9]xxxxxxxxx. This RP will get
match as the output of translated number.
So here is the flow;
Phone -> TP (apply manipulation here) -> RP -> RL -> RG -> Trunk/Gateway
Basically here you've used TP for digit manipulation during outbound call.
2. If we use called party transformation, lets have RP like 91.[2-9]xxxxxxxxx. You also need to
create Called Party Transformation Pattern which can match the configured RP.
Called Party Transformation can look similar to 91.[2-9]xxxxxxxxx. Now you will strip the pre-dot
here under called party transformation which will translate the number to [2-9]xxxxxxxxx. This
transformation pattern is then assigned to trunk/gateway with the help of CSS.
So here is the flow;
Phone -> RP -> RL -> RG -> Trunk/Gateway (apply manipulation here)
- Vivek
See correct answer in context
Correct Answer by Deepak Rawat about 10 months 4 weeks ago
Sanjay,
1) Translation Pattern in simple words change the destination of a call. This is usually used for
incoming calls. However, it can be used for outgoing calls as well but that really does not make
much sense to do since it is actually the Route Pattern that is taking the calls out using
Trunks/Gateways. In your example, you already have a RP created that matches as soon as
agent dials the number 91.7458563514 and call goes out. Now you can simplify this using TP
wherein you can create a TP for lets say 1200 and then you can transform this to
91.7458563514 under Called Party Transform Mask. What it does is that, user does not actually
need to dial the complete long distance number and will just dial 1200 that in turn will change
the called number to 91.7458563514 and will then hit the RP and call will go out. Hence, it is
actually changing the destination.
2)Transformation Pattern simply changes the number mask, your understanding is correct in
that
It does not change the destination, only the display i.e., how ANI and DNIS will looks like once
the transformation had been applied.
Response Acknowledgement