O PCX O Sip T - R8. /R9. /R10. N A: Technical Bulletin Alcatel-Lucent Omnipcx Office

Technical Bulletin Alcatel-Lucent OmniPCX Office
TC1765 ed.04 Release 8,9,10
OMNIPCX OFFICE SIP TRUNKING –

R8.X/R9.X/R10.X NOTEWORTHY ADDRESSES
This document describes a sub-set of system flags (noteworthy addresses) relevant for the configuration of SIP Trunks.
Revision History
Edition 2: February 4, 2014 update of the document
Edition 3: November 26, 2014 update of the document for R10.0
Legal notice:
The information presented is subject to change without notice.
ALE International assumes no responsibility for inaccuracies contained herein.
Copyright © ALE International 2015
Table of contents
1 Introduction/Recommendations ............................................................................................................... 3
2 Flag Section “Other Labels”...................................................................................................................... 4
3 Flag Section “Debug Labels” .................................................................................................................... 7
4 Appendix ................................................................................................................................................ 8
Alcatel-Lucent OmniPCX Office - Release 8,9,10

OmniPCX Office SIP Trunking – R8.x/R9.x/R10.x Noteworthy Addresses TC1765 ed.04
Copyright © ALE International 2015 page 2/24
1 Introduction/Recommendations
- For every version release of the Alcatel OmniPCX Office, there are many Noteworthy Addresses (flag
addresses) that can be modified via OMC.
- It is strongly recommended to only modify those Noteworthy Adresses that are described in Alcatel-Lucent
documentation or Technical Bulletins. When editing a value, special care must be taken to select the right
address as some of the flags may have very similar ID labels.
- When a flag corresponding to a system timer is to be changed, never enter the value FFFFh unless it is
specifically permitted in the doc.
- User flag changes are maintained after a warm reset (unless otherwise stated), but are lost after a cold
reset.
Some flag addresses may need a further system warm reset to validate their value change (when required,
this is indicated in the doc).
- Default value of flags is given in this document for a better illustration of the flag role. However, this default
value may vary depending on the system release considered or the country target selected.
Noteworthy addresses modification with OMC

a) ) In System Miscellaneous" – "Memory Read/Write" choose the appropriate section hosting the address you
want to edit ("Other Labels" / "Timer Labels" or "Debug Labels"). Select the ID label of the address to modify
and click "Details"
b) In the upper part of the window, change the value of the desired byte(s), then click "Modify" and "Write".

2 Flag Section “Other Labels”
No of Default Validity
Label Function description bytes value Remarks
Modify the default codec list priority for outgoing calls :
AlCodLst 1 00
00=G729a-30 / G723.1-30 / G711-30
01 : G723.1-30 / G729a-30 / G711-30
02 : G723.1-30 / G711-30 / G729a-30
03 : G711-30 / G723.1-30 / G729a-30 Valid upto
04 : G711-30 1G729a-30 2G723.1-30 R9.0.
05 : G729a-30 / G711-30 / G723.1-30 Since R9.1,
06 : G711-30 / G729a-30 regular
07 : G729a-30 / G711-30 config in
08 : G711-30 / G723.1-30 OMC
09 : G723.1-30 / 1G711-30
0A : G723.1-30 / G729a-30
0B : G729a-30 / G723.1-30
16 : G711-20 / G729a-20
Value of the Dynamic Paylod for G722 codec

G7222DynPL 1 75
XX : Valeur hexadecimal à convertir en décimal Valid since
6A : 106 Décimal R9.1
75 : 117 Décimal
Value of the Dynamic Paylod for DTMF-RFC4733

DtmfDynPL 1 6A Valid upto
XX : Hexadecimal value to convert in decimal R9.2
Since R10.0,
6A : 106 decimal value
78 : 120 decimal value regular
config in
OMC
Type of CLIP build for a SIP trunk user.
1 22
ExtNuFVoi
00 : only DDI part of the number
02 : zone prefix + Installation No. + DDI No.
11 : subscriber = zone prefix + Installation No. + DDI No.
22 : zone prefix + Installation No. + DDI No.
33 : national code/zone prefix + Installation No. + DDI No.

Preferences of codecs when OXO sends a Re-Invite with Fax G711
2 01 FF
FaxPasCd Offer on detecting fax.
01FF : 8(G711alaw), 0(G711ulaw)

10FF : 0(G711ulaw), 8(G711alaw)
0FFF : 8(G711alaw)
1FFF : 0(G711ulaw)
INVITE with SDP sending on second call leg.

1 00
INVwSDPtrk
00 : INVITE without SDP (Default value)
01 : INVITE with SDP
Force the use of a Codec in case of multiple answers

PrefCodec 2 00 00
00 00 : disabled Note : Is
00 02 : G723 relevant
00 03 : G729 only if
00 04 : G711 A Law “PrefFramin”
00 05 : G711 µ Law is enabled
Force the use of a Framing in case of multiple answers

PrefFramin 1 00 Note : Is
00 = disabled relevant
0A = 10 decimal = 10 ms only if
14 = 20 decimal = 20ms “PrefCodec”
etc… is enabled
Incase of US Transfer, fills the From field of the second call-leg

USalterfrom FROM field can contain the calling party of the the first call-leg 1 00 Since
R10.1/42.01
00 : Normal behavior (Default value)
01 : Second call-leg FROM contains calling party of first call-leg
Controls how the incoming called numbers are managed (private or

VipPuNua public) on the SIP trunk 1 00
00 : Called number based on the SIP Prefixes (both types –Private & Public)
01 : Called number based on the SIP trunk type (only one –Private or
Public)

Enable/disable the external ring back tone simulation. (in case of
SimulIpAlt 183 without RTP). 1 00
00: OXO generates nothing

01: OXO generates a local ring back tone.
Display or not the CNIP (Name) received from the SIP network
SIPInDspNm 1 01
00: CNIP not displayed for both PUBLIC/PRIVATE SIP calls.
01: CNIP displayed for both PUBLIC/PRIVATE SIP calls
03: CNIP displayed for PRIVATE and not displayed for the PUBLIC calls.
Send or not the CNIP (Name) onto the SIP network .

SIPOgDspNm 1 01
Value 00: CNIP not sent for both PUBLIC/PRIVATE SIP Outgoing calls
Value 01: CNIP sent for both PUBLIC/PRIVATE SIP Outgoing calls
Value 03: CNIP sent for PRIVATE calls and not sent for PUBLIC calls

3 Flag Section “Debug Labels”
Re-Invite management in case of multiple codec answers
MultAnsReinv 1 01
01 : Re-invite sent on multiple-codec answer
00 : No Re-invite sent on multiple-codec answer
Manage the inbound « inband” DTMF signaling on a public SIP trunk

SIPdtmfInB 1 00 Valid since
01 : Inband DTMF taken into account R9.2
00 : Inband DTMF ignored
Play or not the Alert tone when we receive the 180 ringing request.
SuprAlerTo 1 00
01 : Don’t play the Alert Tone
00 : Play the Alert Tone
See Appendix
64 4.1
VOIPnwaddr Allows tuning of the SIP protocol on IP Trunks.
for details

4 Appendix
4.1 VOIPnwaddr
Offset 0 Offset 1 Offset 2 Offset 7
Offset 8
Offset 96
Offset 99
4.1.1 Remote SIP Port
Offset 0 & 1 On 2 bytes Default value is 00 00 Obsolete since R6.0 – use OMC parameter
Description
This value sets the remote port number used by SIP trunk. The outgoing Invite will be sent to the defined port
value . If the value is 0, the default SIP will be used (5060).

4.1.2 Privacy Level
Offset 2 On 1 byte Default value is 00 /
Description
Set the OXO privacy policy when CLIR is used.

If identity presentation of user 1234 is restricted, the From field of an outgoing INVITE from the OXO will be :
- 00 : From: sip:anonymous@anonymous.invalid
- 01 : From: sip:1234@LocalDomain
4.1.3 To as Req-URI
Description
This flag is used for building the “To” field for outgoing SIP calls. By default, the “To” field is the same as
“Request URI” except for private forward. In this case, the “To” field is the diverting number (the
destination of the first call). To fix the “To” field to be identical as “Req-UR”I, the value must be set to 1.
- 00 : INVITE 1234@LocalDomain & To: sip:5678@LocalDomain

- 01 : INVITE 1234@LocalDomain To: sip:1234@LocalDomain
Note: if called is public, the To field will be always the same as the Req URI.
4.1.4 Session Timer
Offset 4 & 5 On 2 bytes Default value is 00 00 Obsolete since R10.0 – use OMC parameter
Description
The session timer is the delay parameter specified by RFC4028. In the context of each call, a keep alive
(session refresh) is performed at 50% of the period specified by this variable : it consists in a Re-Invite or an
Update. If no refresh is performed / successful at the end of this timer, the call is released. The unit is the
minute.
- 0000 : Default timer, which is 720 minutes (43200 sec = 12 hours),

- FFFF : Disable Session Timer.
- xxxx : Any other value specifies the period in minutes unit.

4.1.5 Don’t use DNS SRV unreachable proxy list
Offset 6 On 1 byte Default value is 00 Obsolete since R10.0 – use OMC parameter
Description
DNS SRV makes use of a quarantine list to memorize the unreachable proxies : it optimizes overflow by not
trying known unreachable proxies.
- 00 : Use unreachable proxy list.

- 01 : Don’t use unreachable proxy list.
4.1.6 T.38 FAX inhibition
Description
FAX are transmitted with T.38 protocol by default
- 00 : T.38 FAX protocol : (use the OMC/Gateway parameter to set up the FAX transmission mode)
- 01 : No T.38 FAX protocol (use the OMC/Gateway parameter to set up the FAX transmission mode)
- 02: No T38 FAX detection when G711mode is configured.
4.1.7 NAT Keep Alive for DNS SRV
Offset 8 & 9 On 2 bytes Default value is 00 00 /
Description
In context of DNS-SRV and OXO placed behind a NAT/Firewall OXO will send OPTION messages at 75% of
this delay to avoid the NAT connection to be removed.
- 00 00 : no NAT Keep Alive.

- 02 58 : Timer is 0258 Hex = 600 decimal = 600 seconds for ie
- xx xx : NAT Keep Alive is enabled for DNS SRV rules, and specifies the NAT connection duration

4.1.8 SIP Trunk Signaling Source Port
Offset 10 & 11 On 2 bytes Default value is 00 00 See notes
Description
Force the OXO’s source port for SIP signalling. This parameter is associated to both UDP and TCP transport.
- 00 00 : The source port is SIP stack dependent, it uses either a dynamically allocated source port or a
static 5060 source Port (Sip stack version dependant).
- 13 C4 : Hex value is 5060 Decimal value = So port is 5060 for ie.
- xx xx : Use this port value as source for SIP signalling
Notes :
- Since R710 / Fixed to 5060, don’t use anymore the flag.
- Since R910 / Must be configured in OMC via « Voice Over IP/Gateway” parameter “SIP trunk
signal source port”
4.1.9 Rport activation
Description
This parameter permits to disable the SIP stack rport.
- 00: rport is activated

- 01: rport is disabled
4.1.10 Unavailable Voip Cause
Description
If Call server sends destination out of order as release cause :
- 00: SIP gw sends 404 Not found to the network.

- 01: SIP gw sends 502 Bad gateway.

4.1.11 Name display with CLIR
Description
If the SIP header of an incoming INVITE contains From: ”name” <anonymous@anonymous.invalid>.
- 00: “name” is not displayed.

- 01: “name” is displayed.
4.1.12 Registration identifier
Offset 15 On 1 byte Default value is 00 Value 8 introduced since R8.2 only/ Obsolete
since R10.0 – use OMC parameter
Description
Use of the AOR in Invite, if :
- 00: No AOR. DID number of user is sent along with the INVITE message
- 01: Use the registration AOR in P-asserted id for outgoing INVITE.
- 02: Use the registration Contact in outgoing INVITE’s Contact header.
- 03: Use Registration AOR in P-asserted id and registration Contact in INVITE’s Contact (cumulate 01&02).
- 04: Use the registration AOR in P-preferred id for outgoing INVITE.
- 08: Use the registration AOR in P-asserted id and contact header for all INVITE request sent (Re-Invite).
- 10: (Since R900) Use the registration AOR in From and contact header for all INVITE request sent (Re-
Invite).
4.1.13 P-Prefered id
Description
Use of P_prefered & P_asserted headers
- 00: Outgoing calls: no P-Pi / Incoming Calls: Process P-Pi only if P-AI is absent.
- 01: Outgoing Calls: add a P-Pi with user identity (DID) / Incoming calls: Process P-Pi only if P-Ai is absent.
- 02: Outgoing Calls: no P-Pi / Incoming calls: Process P-Pi in priority.
- 03: Outgoing Calls: add a P-Pi with user identity (DID) / Incoming calls: Process P-Pi in priority.
- 04: Outgoing Calls: no P-Pi. No P-Ai / Incoming calls: Process From header in priority.
- 05: Outgoing Calls: add a P-Pi with user identity (DID). No P-Ai / Incoming calls: Process From

header in priority.
- 08: (Since R820) Outgoing Calls: add a P-Ai with user identity (DID). No P-Pi / Incoming calls: Process P-
Pi only if P-AI is absent. Called number displayed on calling party : Display To, if neither P_Pi nor P-Ai in
180 & 200.
- 10: (Since R900 & if VOIPnwaddr+15=”10”) Outgoing Calls: add a From with user identity (DID). /
Incoming calls: Process From.
- 20: (Since R920) Outgoing calls: no P-Pi / Incoming Calls: Process P-Pi only if P-AI is absent.
Called number displayed on calling party : Keep the dialled digits
Following table describes the noteworthy values translated in OMC data after a system migration
Calling Preferred Identity Connected Preferred Identity

Value
Incoming Outgoing Outgoing
Outgoing Header List
header list P-Pi P-Ai
P-Asserted-id
P-Asserted-id
P-Preferred-id
00 P-Preferred-id Disabled Enabled Enabled
Contact
From
To
P-Asserted-id
P-Asserted-id
P-Preferred-id
01 P-Preferred-id Enabled Enabled Enabled
Contact
From
To
P-Asserted-id
P-Preferred-id
P-Preferred-id
02 P-Asserted-id Disabled Enabled Enabled
Contact
From
To
P-Asserted-id
P-Preferred-id
P-Preferred-id
03 P-Asserted-id Enabled Enabled Enabled
Contact
From
To
From P-Asserted-id
P-Asserted-id P-Preferred-id
04 Disabled Enabled Enabled
P-Preferred-id Contact
To
From P-Asserted-id
05 Enabled Disabled Enabled
To
P-Asserted-id
P-Asserted-id
P-Preferred-id
08 P-Preferred-id Disabled Enabled Enabled
To
From
Contact
From P-Asserted-id
10 Disabled Enabled Enabled
To
P-Asserted-id
P-Asserted-id
P-Preferred-id
20 P-Preferred-id Disabled Enabled Disabled
Contact
From
To

4.1.14 Support of the Update Method
Description
Enable/disable the support of the Update method.
- 00: OXO is allowed to send or receive “UPDATE” method (default value)

- 01: OXO is not allowed to send or receive “UPDATE” method
4.1.15 TCP switching
Description
Enable/disable the UDP to TCP switching.
- 00: OXO is allowed to switch from UDP to TCP.

- 01: OXO is not allowed to switch from UDP to TCP
Note : Value 01 sets the flag “Do Not Fragment” to false (in IP Layer)

4.1.16 Send No-Signal
Offset 19 On 1 byte Default value is 00 Value 02 introduced in R9.0/ Obsolete since

R10.0 – use OMC parameter
Description
Send no-signal message for outgoing fax call.
- 00: OXO does not send no-signal message

- 01: OXO send no-signal messages for outgoing fax call
- 02: (Since R900) OXO sends CNG messages for outgoing fax call
4.1.17 History-info
Description
Enable/disable the support of History-Info header (RFC4244) at transmit and receive.
- 00: History-Info processing is enabled

- 01: History-Info processing is disabled
4.1.18 Optimization of authentication
Description
Enable/disable the support of Optimization of Authentication .
- 00: Optimization of Authentication is enabled

- 01: Optimization of Authentication is disabled

4.1.19 DNS-A resolution
Description
Possibility for OXO to enable the Force DNS A resolution.
- 01: Force OXO to make DNS A resolution instead of DNS SRV resolution.
- 00: DNS_SRV is enabled and DNS A resolution is disabled for OXO.However, OXO can make DNS A
resolution only when the request for SRV records resulted in “no such record” response.
4.1.20 Registration check
Description
Enable registration check before routing requests (register and non-register) to the outbound proxy.
- 00: No checking for presence of a valid registration is done before routing requests.
- 01: Presence of a valid registration is checked before routing non-register requests to the outbound proxy.
If a valid registration to the outbound proxy doesn’t exist, a REGISTER request is triggered for
registration.
4.1.21 V21 Jitter Buffer
Description
Tune the depth of the fax V21 jitter buffer. The unit is the millisecond.
- 00 00 : Default V21 jitter buffer depth applies, i.e 240ms.

- 02 00 : Hex value is 512 Decimal value = 512ms.
- xx xx : Timer value in ms.

4.1.22 T4 Jitter Buffer
Description
Tune the depth of the fax T4 jitter buffer. The unit is the millisecond.
- 00 00 : Default T4 jitter buffer depth applies, i.e 240ms.

- 02 00 : Hex value is 512 Decimal value = 512ms.
- xx xx : Timer value in ms.
4.1.23 Alert Message
Description
Send 180 Ringing and 183 Session Progress, when 180 Ringing or 183 Session Progress with SDP is received,
with the delay (~1s) after 100 Trying is received in transit case.
- 00: In transit case, if 180 Ringing or 183 Session Progress is received with SDP with a delay (~1s) after
receiving 100 Trying, OXO sends Session Progress (183) with SDP to another call leg.
- 01: In transit case, if 180 Ringing or 183 Session Progress is received with SDP with a delay (~1s) after
receiving 100 Trying, OXO sends Alert message (180 Ringing), which is followed by Session Progress
(183) with SDP to another call leg.
4.1.24 SIP Capabilities
Description
Display the sip capabilities in the contact header of the register request.
- 00: Presence of a sip_capabilities in the contact header of the register request.

- 01: No Presence of a sip_capabilities in the contact header of the register request.

4.1.25 Source Port for SIP Phones
Offset 30 & 31 On 2 bytes Default value is 00 00 Introduced in R8.1/ Obsolete since R9.1 – use
OMC parameter
Description
Force the OXO’s source port for SIP Phone signalling in UDP and TCP.
- 00 00 : The source port is 5059.

- 13 C1 : Hex value is 5057 Decimal value = So port is 5057 for ie.
- xx xx : Use this port value as source for SIP signalling
4.1.26 Reserved
Offset 32 & 33 On 2 bytes Default value is 00 00 Reserved
4.1.27 T38 CED
Description
Send simulated T38 ced message to the network.
- 00: Simulated T38 ced message will not be sent to the network
- 01: Simulated T38 ced message will be sent to the network
4.1.28 IP Authentication
Offset 35 On 1 byte Default value is 00 Introduced in R8.1/ Obsolete since R10.0 –

use OMC parameter
Description
Enable authentication of incoming call based on IP address for DNS enabled ARS lines.
- 00: Source IP address is not checked for incoming calls associated to DNS ARS lines.
- 01: Source IP address is checked for incoming calls associated to DNS ARS lines.

4.1.29 Session Timer for SIP Phones
Offset 36 & 37 On 2 bytes Default value is 00 00 Introduced in R8.2
Description
The session timer is the delay parameter specified by RFC4028. In the context of each call, a keep alive
(session refresh) is performed at 50% of the period specified by this variable : it consists in a Re-Invite or an
Update. If no refresh is performed / successful at the end of this timer, the call is released. The unit is the
minute.
- 00 00 : Default timer, which is 1800 seconds (30mn),

- 2A 30 : Maximum value (3 Hours)
- xx xx : Any other value specifies the period in seconds unit.
4.1.30 Registration Triggering causes
Offset 38 to 57 On 20 bytes Default is Introduced in R820

00 00
00 00
00 00
00 00
00 00
00 00
00 00
00 00
00 00
00 00
Description
This parameter is an array of ten elements (of 2 bytes) : Each element is a failure response code (4xx or 5xx
or 6xx). This list of cause allows the triggering of registration when an INVITE requests fails with an error
response code, in the context of a proxy failover using DNS.
- 00 00 : Empty element
- 01 93 : Means 403 Decimal, is 403 Forbidden.
- xx xx : SIP failure response code (4xx, 5xx, 6xx) coded in hexadecimal..

4.1.31 P-early-media Activation
Offset 58 On 1 byte Default value is 00 Introduced in R9.0/ Obsolete since R10.0 –

use OMC parameter
Description
This parameter offers the ability to enable the control of early media flow through p-early-media
- 00 : P-Early-Media is disabled, no control of early media flow.

- 01 : P-Early-Media is enabled, control of early media flow.
4.1.32 G711A Silence Suppression
Offset 59 On 1 byte Default value is 00 Introduced in R900
Description
Disable the silence suppression for G711A codec in both Sip Phone and SIP Trunk gateway.
- 00 : Silence Suppression (VAD) value configured in OMC->Voice Over IP->VoIP: Parameters will be used
for G711A codec.
- 01 : Silence Suppression (VAD) is disabled for G711A Codec alone.
4.1.33 FAX ECM
Description
disable the FAX ECM mode on SIP or H.323 trunking.
- 00 : Fax ECM is enabled.

- 01 : Fax ECM is disabled.

4.1.34 Session Timer for SIP Companion
Offset 61 & 62 On 2 bytes Default value is 00 00 Introduced in R910
Description
The session timer is the delay parameter specified by RFC4028. In the context of each call of a SIP
companion, a keep alive (session refresh) is performed at 50% of the period specified by this variable : it
consists in a Re-Invite or an Update. If no refresh is performed / successful at the end of this timer, the call is
released. The unit is the minute.
- 00 00 : Default timer, which is 90 seconds

- 2A 30 : Maximum value (10800 sec = 3 Hours)
- xx xx : Any other value specifies the period in seconds unit.
4.1.35 Username in the initial REGISTER
Offset 63 1 Octet Default value is 00 Introduced in R9.2/ Obsolete since R10.0 –

use OMC parameter
Description
Enable/disable the sending of the Username (private user identity) in the first REGISTER.
Compliance with 3GPP TS 24.229 specification.
- 00 : No authentication field with the Username in the first REGISTER sent.

- 01 :The first REGISTER sent, includes an authentication field with the Username .
4.1.36 SIP OPTION request to DNS cache IP adresses
Offset 64 1 Octet Default value is 00 Introduit en R900
Description
Enable/disable the ability to send SIP OPTION requests to all IP adresses present in the DNS cache.
This parameter will work if DNS_SRV mode is activated and the «NAT Keep Alive for DNS SRV» flag value is
not “00 00”.
- 00 : No OPTION request sent to ALL DNS cache IP adresses.

- 01 : OPTION request sent to ALL DNS cache IP adresses.

4.1.37 Special character truncation for called number (outgoing call)
Offset 65 1 Octet Default value is 00 Introduced in R8.2
Description
Enable special char truncation in outcalled number.
- 00 : NO truncation of special character and further digit(s)/char(s) in outCcalled numbers.

- 01 : HASH(#) char and further digit(s)/char(s) are truncated in outCalled numbers
4.1.38 Disable sending of UPDATE without SDP when connected identity changes
Description
If UPDATE method allowed on OXO, then UPDATE without SDP messages are sent by
OXO when connected identities change. Some core networks may not support the reception
of such messages and it involves extra traffic that is not desired.
With this noteworthy address, it‟s possible to decide to send or not this kind of UPDATE
messages ( RFC4916 configuration ).
- 00 : sending of these UPDATE messages is enabled.

- 01 : sending of these UPDATE messages is disabled
4.1.39 Process To header for incoming call routing in SIP Trunk gateway
Description
Enable/disable the ability to option to process SIP To header in priority for all the incoming INVITE
requests. Hence, Call routing will be based on SIP To header only.
- 00 : OXO will route calls based on the Request URI. If „user‟ field of To header and Request URI are
different then call will be routed based on Request URI and the set will display diverted number (i.e. „user‟
field of To Header).
- 01 : OXO will process all the incoming INVITE request based on SIP To header. If Request URI and To
header are different then Call will be routed based on TO header only and the set will display the number
present in the TO header

4.1.40 Remove maxptimer attribute from all INVITE offers
Offset 69 1 Octet Default value is 00 Introduced in R920/069.01
Description
Option to remove maxptime attribute present in the SDP of all INVITE offers from OXO in SIP Trunk Gateway.
This maxptime attribute will be removed only when codec framing is forced in ARS and not when “Default” is
selected.
- 00 : maxptime attribute will be present in all INVITE SDP offers.

- 01 : maxptime attribute will be absent in initial INVITE SDP offer.
- 02 : maxptime attribute will be absent in all INVITE SDP offers
4.1.41 Media management after Re-INVITE
Offset 70 1 Octet Default value is 00 Introduced in R100/048.01
Description
Option to modify the FAX or AUDIO switching when receiving AUDIO and FAX after a RE-INVITE.
- 00 : 1) Call treated as a FAX when OXO receives Audio as First MEDIA and Fax as next MEDIA.
2) Call treated as a FAX when OXO receives FAX as the first MEDIA with/without Audio as next MEDIA.
- 01 : 1) Call treated as an AUDIO when OXO receives Audio as First MEDIA and Fax as next MEDIA.
2) Call treated as a FAX when OXO receives FAX as the first MEDIA with/without Audio as next MEDIA.
NOTE : The behavior associated to value 01 is the R920 default mode.

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TC1765 ed.04 Release 8,9,10

OMNIPCX OFFICE SIP TRUNKING –

Alcatel-Lucent OmniPCX Office - Release 8,9,10

Noteworthy addresses modification with OMC

Alcatel-Lucent OmniPCX Office - Release 8,9,10

Value of the Dynamic Paylod for G722 codec

Value of the Dynamic Paylod for DTMF-RFC4733

Alcatel-Lucent OmniPCX Office - Release 8,9,10

01FF : 8(G711alaw), 0(G711ulaw)

INVITE with SDP sending on second call leg.

Force the use of a Codec in case of multiple answers

Force the use of a Framing in case of multiple answers

Incase of US Transfer, fills the From field of the second call-leg

Controls how the incoming called numbers are managed (private or

Alcatel-Lucent OmniPCX Office - Release 8,9,10

00: OXO generates nothing

Send or not the CNIP (Name) onto the SIP network .

Alcatel-Lucent OmniPCX Office - Release 8,9,10

Manage the inbound « inband” DTMF signaling on a public SIP trunk

Alcatel-Lucent OmniPCX Office - Release 8,9,10

4.1.1 Remote SIP Port

Alcatel-Lucent OmniPCX Office - Release 8,9,10

Offset 2 On 1 byte Default value is 00 /

Set the OXO privacy policy when CLIR is used.

Offset 3 On 1 byte Default value is 00 /

- 00 : INVITE 1234@LocalDomain & To: sip:5678@LocalDomain

4.1.4 Session Timer

- 0000 : Default timer, which is 720 minutes (43200 sec = 12 hours),

Alcatel-Lucent OmniPCX Office - Release 8,9,10

- 00 : Use unreachable proxy list.

4.1.6 T.38 FAX inhibition

Offset 7 On 1 byte Default value is 00 /

FAX are transmitted with T.38 protocol by default

4.1.7 NAT Keep Alive for DNS SRV

Offset 8 & 9 On 2 bytes Default value is 00 00 /

- 00 00 : no NAT Keep Alive.

Alcatel-Lucent OmniPCX Office - Release 8,9,10

Offset 10 & 11 On 2 bytes Default value is 00 00 See notes

4.1.9 Rport activation

Offset 12 On 1 byte Default value is 00 /

This parameter permits to disable the SIP stack rport.

- 00: rport is activated

4.1.10 Unavailable Voip Cause

Offset 13 On 1 byte Default value is 00 /

If Call server sends destination out of order as release cause :

- 00: SIP gw sends 404 Not found to the network.

Alcatel-Lucent OmniPCX Office - Release 8,9,10

Offset 14 On 1 byte Default value is 00 /

If the SIP header of an incoming INVITE contains From: ”name” <anonymous@anonymous.invalid>.

- 00: “name” is not displayed.

4.1.12 Registration identifier

Use of the AOR in Invite, if :

Use of P_prefered & P_asserted headers

Alcatel-Lucent OmniPCX Office - Release 8,9,10

Calling Preferred Identity Connected Preferred Identity

Alcatel-Lucent OmniPCX Office - Release 8,9,10

Enable/disable the support of the Update method.

- 00: OXO is allowed to send or receive “UPDATE” method (default value)

4.1.15 TCP switching

Enable/disable the UDP to TCP switching.

- 00: OXO is allowed to switch from UDP to TCP.

Alcatel-Lucent OmniPCX Office - Release 8,9,10