SIP set up on LDK100

[Digitalcoms]

Hie everyone

I got SIP trunks from public provider that i want to connect on LDK100. I have gone through threads on this forum and i am not getting anywhere

how can i send .USR file to someone to help me on the configuration. i am getting lost somewhere.

plse someone help!!!

 

[Digitalcoms]

hie doktor
can u please help me setup SIP on iPLDK100. I followed threads on this forum and i still can't register the lines. i dumped .USR file on this link

http://www.mediafire.com/?k95molegqsmpl19

 

[Digitalcoms]

Sorry the Firmware vers is GS90P-3.9Ah and PCADMIN 3.9Dc

No password

 

[doktor]

An LDK-300 VOIBE board is needed. I will have a look...

///doktor

 

[doktor]

Is it a LDK-300 VOIBE (has to be "E") in SLOT=5??

///doktor

 

[Digitalcoms]

No doktor. It is VOIB in slot 5 with 4 VOIU units to give 8 channels. Firmware version 4.2Aa

 

[doktor]

I think that only VOIBE can run SIP.
VOIB/VOIU can run only H.323.

///doktor

 

[Digitalcoms]

thank you doktor. i will get voibe. did u check the .USR file. are the sttings correct in case i get the VOIBE.

again thanks doktor for your help. will advise when i get voibe.

 

[doktor]

Hi there. I got your mail, but please re-send it.
Also check with your vendor if VOIBE supports SIP. (I am sure, but it is better to check again.)

///doktor

 

[Digitalcoms]

thanx doktor. here is the link for the .USR file.
Meanwhile i check specs for VOIB & VOIBE

http://www.mediafire.com/?k95molegqsmpl19

 

[doktor]

OK.
First check this link to download the newewat docs:

http://www.artcom.ru/download/?TYPE=6&MODEL=507#

(ipLDK-100)

Find this manual - in English. It is number 4 from top.

Руководство по программированию АТС ipLDK версий 3.9 (англ. язык)
(.rar ~4.49 Mb)

Check page 214-219 for the SIP operation.
I will paste oi in the next "post".

///doktor

 

[doktor]

Here is the SIP programming overview. To go to details you need the programming manual:

2.20.1 Incoming Call
Description
ipLDK system can receive incoming call by two methods. Two methods are TRUNK mode and REGISTER mode. TRUNK mode finds user agent IP address by proxy sever configuration. REGISTER mode receives incoming call using REGISTER method(2.21.3). The rule of SIP Incoming Call is same as 2.1 How To Get Incoming Call.


Operation
Example
 Trunk mode : Make Station 1000~1009 receive incoming call from SIP server(sip.trunk.com)
1. Set Station number as 1000~1009 at ADMIN 105.
2. Set CO Service type as DID at ADMIN 140.
3. Set COLP Table Index as 00 at ADMIN 143.
4. Set CLIP Table Index as 00 at ADMIN 143.
5. Set Call Type as SUBSCRIBER at ADMIN 143.
6. Set DID CONV Type as 1 at ADMIN 143.
7. Set ISDN Enblock Send as ON at ADMIN 143.
8. Set Networking CO Line Type as SIP at ADMIN 322.
9. Set VOIB Mode as SIP at ADMIN 340.
10. Set Proxy Server Address as sip.trunk.com at SIP Attribute 1.
11. Set Domain as sip.trunk.com at SIP Attribute 1.
=> When there’s an incoming call, the Station 1000~1009 can receive call by SIP URI number.

 REGISTER mode : Make Station 1000 receive incoming call from SIP server(sip.reg.com)
1. Set Station number as 1000 at ADMIN 105.
2. Set SIP User ID Table Index as 1 at SIP Attribute 2.
3. Set CO Service type as DID at ADMIN 140.
4. Set COLP Table Index as 00 at ADMIN 143.
5. Set CLIP Table Index as 00 at ADMIN 143.
6. Set Call Type as SUBSCRIBER at ADMIN 143.
7. Set DID CONV Type as 1 at ADMIN 143.
8. Set ISDN Enblock Send as ON at ADMIN 143.
9. Set Networking CO Line Type as SIP at ADMIN 322.
10. Set VOIB Mode as SIP at ADMIN 340.
11. Set Proxy Server Address as sip.reg.com at SIP Attribute 1.
12. Set Domain as sip.trunk.com at SIP Attribute 1.
13. Set User ID as 1000@sip.reg.com at SIP Attribute 2.
14. Set Contact Number as 1000 at SIP Attribute 2.
15. Set User ID Register as ON at SIP Attribute 2.
16. Set User ID Usage as ON at SIP Attribute 2.
=> When there’s an incoming call, the Station 1000 can receive call by registration.


Condition
1. For using SIP CO Line, VOIB Mode should be SIP or Dual mode.
2. Both NOMAL and DID CO service type supported.
3. MSN service applied in SIP CO Line.
4. CLIP and COLP are applied in SIP CO Line.
5. Mobile Extension features applied by SIP CO Line.

Admin Programming
 CO Service Type 4.2.1 (PGM 140)
 COLP Table Index 4.2.4.1 (PGM 143 – FLEX 1)
 CLIP Table Index 4.2.4.2 (PGM 143 – FLEX 2)
 Call Type 4.2.4.3 (PGM 143 – FLEX 3)
 DID CONV Type 4.2.4.4 (PGM 143 – FLEX 4)
 DID Remove Number 4.2.4.5 (PGM 143 – FLEX 5)
 ISDN Enblock Send 4.2.4.6 (PGM 143 – FLEX 6)
 CLIP/COLP Table 4.8.2 (PGM 201)
 Networking CO Line Type 4.12.3.2 (PGM 322 – FLEX 2)
 VOIB Mode 4.13.1.12 (PGM 340 – FLEX 12)
 SIP Attribute 1 (PGM 500)
 SIP Attribute 2 (PGM 501)



2.20.2 Outgoing Call
Description
ipLDK system makes Outgoing Call same as rule of 2.2 How To Access Outgoing Call.


Operation
Example
ADMIN setting is same as 2.12.1 Incoming Call Example.
1. Lift Handset or press the [MON] button.
2. Press the desired CO Line, {POOL} button, or {LOOP} button.
3. Or, dial the individual CO Line access code, CO group access code, or the first CO Line access code from the accessible group.
4. Press destination number and press ‘#’. If not press ‘#’, call sent by Enblock Digit Timer.

Condition
1. For using SIP CO Line, VOIB Mode should be SIP or Dual mode.
2. TRUNK and REGISER mode use same method.
3. Both NOMAL and DID CO service type supported.
4. MSN service applied in SIP CO Line.
5. CLIP and COLP are applied in SIP CO Line.
6. Mobile Extension features applied by SIP CO Line.
7. SIP CO Line must use Enblock Send feature.

Admin Programming
 CO Service Type 4.2.1 (PGM 140)
 COLP Table Index 4.2.4.1 (PGM 143 – FLEX 1)
 CLIP Table Index 4.2.4.2 (PGM 143 – FLEX 2)
 Call Type 4.2.4.3 (PGM 143 – FLEX 3)
 DID CONV Type 4.2.4.4 (PGM 143 – FLEX 4)
 DID Remove Number 4.2.4.5 (PGM 143 – FLEX 5)
 ISDN Enblock Send 4.2.4.6 (PGM 143 – FLEX 6)
 Enblock Digit Timer 4.5.3.10 (PGM 182 – FLEX 10)
 CLIP/COLP Table 4.8.2 (PGM 201)
 Networking CO Line Type 4.12.3.2 (PGM 322 – FLEX 2)
 VOIB Mode 4.13.1.12 (PGM 340 – FLEX 12)
 SIP Attribute 1 (PGM 500)
 SIP Attribute 2 (PGM 501)


2.20.3 Register
Description
The REGISTER method is used by a user agent notify a SIP network of its current Contact URI (IP address) and the URI that should have requests routed to this CONTACT. SIP registration bears some similarity to cell phone registration on initialization. Registration is not required to enable a user agent to use a proxy server for outgoing calls. It is necessary, however, for a user agent to register to receive incoming calls from proxies that serve that domain unless some non-SIP mechanism is used by the location service to populate the SIP URIs and Contacts of end-points.


Operation
Example
1. Set Station number as 1000 at ADMIN 105.
2. Set SIP User ID Table Index as 1 at SIP Attribute 2.
3. Set CO Service type as DID at ADMIN 140.
4. Set COLP Table Index as 00 at ADMIN 143.
5. Set CLIP Table Index as 00 at ADMIN 143.
6. Set Call Type as SUBSCRIBER at ADMIN 143.
7. Set DID CONV Type as 1 at ADMIN 143.
8. Set ISDN Enblock Send as ON at ADMIN 143.
9. Set Networking CO Line Type as SIP at ADMIN 322.
10. Set VOIB Mode as SIP at ADMIN 340.
11. Set Proxy Server Address as sip.reg.com at SIP Attribute 1.
12. Set Domain as sip.trunk.com at SIP Attribute 1.
13. Set User ID as 1000@sip.reg.com at SIP Attribute 2.
14. Set Contact Number as 1000 at SIP Attribute 2.
15. Set User ID Register as ON at SIP Attribute 2.
16. Set User ID Usage as ON at SIP Attribute 2.
=> Station 1000 send REGISTER message after reset system.

Condition
1. If you set User ID Provision as Register, REGISTER method will be sent after initialization.
2. One SIP User ID cab be shared many extensions. In this method, all extensions can make call with only one registration.

Admin Programming
 CO Service Type 4.2.1 (PGM 140)
 COLP Table Index 4.2.4.1 (PGM 143 – FLEX 1)
 CLIP Table Index 4.2.4.2 (PGM 143 – FLEX 2)
 Call Type 4.2.4.3 (PGM 143 – FLEX 3)
 DID CONV Type 4.2.4.4 (PGM 143 – FLEX 4)
 DID Remove Number 4.2.4.5 (PGM 143 – FLEX 5)
 ISDN Enblock Send 4.2.4.6 (PGM 143 – FLEX 6)
 Enblock Digit Timer 4.5.3.10 (PGM 182 – FLEX 10)
 CLIP/COLP Table 4.8.2 (PGM 201)
 Networking CO Line Type 4.12.3.2 (PGM 322 – FLEX 2)
 VOIB Mode 4.13.1.12 (PGM 340 – FLEX 12)
 SIP Attribute 1 (PGM 500)
 SIP Attribute 2 (PGM 501)

2.20.4 Private Extension
Description
This Private Extension enable a network of trusted SIP servers to assert the identity of end users or end systems, and to convey indications of end-user requested privacy. The use of these extensions is only applicable inside a ‘Trust Domain’ as defined in Short term requirements for Network Asserted Identity. Nodes in such Trust Domain are explicitly trusted by its users and end-system to publicly assert the identity of each party, and to be responsible for withholding that identity outside of the Trust Domain when privacy is requested.


Operation
Example
1. Set Station number as 1000 at ADMIN 105.
2. Set SIP User ID Table Index as 1 at SIP Attribute 2.
3. Set CO Service type as DID at ADMIN 140.
4. Set COLP Table Index as 00 at ADMIN 143.
5. Set CLIP Table Index as 00 at ADMIN 143.
6. Set Call Type as SUBSCRIBER at ADMIN 143.
7. Set DID CONV Type as 1 at ADMIN 143.
8. Set ISDN Enblock Send as ON at ADMIN 143.
9. Set Networking CO Line Type as SIP at ADMIN 322.
10. Set VOIB Mode as SIP at ADMIN 340.
11. Set Proxy Server Address as sip.reg.com at SIP Attribute 1.
12. Set Domain as sip.trunk.com at SIP Attribute 1.
13. Set Asserted ID(or Remote Party ID) Usage as ON at SIP Attribute 1.
=> When there’s an outgoing call, the INVITE message include P-Asserted-Identify(or Remote-Party-ID) header field.

Condition
1. P-Asserted-Identity and Remote-Party-ID can’t used at the same time.
2. P-Asserted-Identity enable ‘Id’ Privacy Type at SIP Attribute 2 User Privacy.
3. Remote-Party-Id enable ‘Privacy’ at SIP Attribute 2 User Privacy. If this value is set, the INVITE message include ‘Privacy=full’. If this value is off, the INVITE message include ‘Privacy=off’.
4. This Private Extension makes URI by the Station(not by User ID at SIP Attribute 2).

Admin Programming
 CO Service Type 4.2.1 (PGM 140)
 COLP Table Index 4.2.4.1 (PGM 143 – FLEX 1)
 CLIP Table Index 4.2.4.2 (PGM 143 – FLEX 2)
 Call Type 4.2.4.3 (PGM 143 – FLEX 3)
 DID CONV Type 4.2.4.4 (PGM 143 – FLEX 4)
 DID Remove Number 4.2.4.5 (PGM 143 – FLEX 5)
 ISDN Enblock Send 4.2.4.6 (PGM 143 – FLEX 6)
 Enblock Digit Timer 4.5.3.10 (PGM 182 – FLEX 10)
 CLIP/COLP Table 4.8.2 (PGM 201)
 Networking CO Line Type 4.12.3.2 (PGM 322 – FLEX 2)
 VOIB Mode 4.13.1.12 (PGM 340 – FLEX 12)
 SIP Attribute 1 (PGM 500)
 SIP Attribute 2 (PGM 501)


2.20.5 SIP Session Timer
Description

1. Session Expire Timer.
The time at which an element will consider the call timed out, if no successful INVITE transaction or UPDATE transaction occurs beforehand. This value is inserted into every INVITE and UPDATE transaction in the Session-Expires header unless it was configured to zero. A zero session Expires means that the Session Timer feature is turned off. If the “timer” option tag is not part of the supported list, the session Expires value will be ignored.

2. minSE.
The minimum value for the session interval that the application is willing to accept.
If the application does not set this parameter, the minSE value is set to the default value of 90 seconds according to the Session Timer RFC. Also, the Min-SE header will not be present in the sent requests (except for a request, following a 422 response). However, if the application sets this parameter to 90 or any other value, the Min-SE header will appear in any sent request

Operation

Condition

Admin Programming
 SIP Attribute 1 (PGM 500)
 SIP Attribute 2 (PGM 501)



///doktor

 

[doktor]

Checked out if VOIB can run SIP, but it cannot!

This remark is from the Compabililty Table aso found on the mentioned web site:
From F/W Version 4.3Ae, NAT and Gatekeeper is available but SIP is not available.

You need to run VOIBE with s/w 2.1Dc

///doktor

 

[Digitalcoms]

thank you doktor. i will get voibe and advise you progress.