Third party SIP Phone unable to register on the SLG of CS1000

[nortavaya]

Hi all,

We try to register a third party SIP Phone to the CS1K (through SLG)but with no succes, we did all required configuration on the CS1K side and SIP Phone side as follow:

1. Enable a SIP Third Party Licence on the CS1K System
2. Enable SLG on the Node IP
3. Configure SLG settings on the NodeIP (SIP Domain, SLG port, MO IP address, TCP...etc)
4. Create DCH,Zone, RDB and members
5. Use the existing AML 16 and VAS 16 (they are already used for AACC and enabled, can we use the same, or I should create new VAS/AML ?)
6. Create SIPL phone as UEXT type

Then, when I try to register the SIP Phone, I got error on the phone saying failed to register

How I can check if my SLG is up and running ? note that we have Signaling Server Linux based

Also, how I can enable realtime SIP Traces if SLG in order to see what happens and SIP messages exchanges ?


Thank you in advance

 

[nortavaya]

Hi,

It works only when I redial from missed call list of the SIP Phone

On the DCH I got this:

DCH 100 IMSG SETUP REF 00008181 CH 100 0 0 0 TOD 10:52:24
CALLING #:58117 NUM PLAN: NUM UNKNOWN/UNKNOWN (UNKNOWN) ===> DN Of the SIP Phone
CALLED #:2958117 NUM PLAN: PRIVATE/ABBREVIATED (CDP) ==> HOT U UADN

I didn't understand why on the monitor we see the configured number as "HOT U UADN" of the SIP UEXT, however we call another extension ??!!

 

[mckinneyj]

Try and check your MSEC prompt in the config record. should be off or no, I forgot which it wanted

 

[roc221]

what type of 3rd party phone are you connecting ????

 

[nortavaya]

It is conference SIP Phone, type Polycom RealPresence Trio 8800

I tried to turn On/Off MSEC, but still the same issue

 

[roc221]

i have an CS1k 7.5 and going by the dev lab doc they had for polycom duo phone was . wrong in so many ways , what doc are you using to programm the phones

 

[nortavaya]

Yes, I use the document for duo phone with CS1K, see the link below:

https://www.devconnectprogram.com/fileMedia/download/ecfb3c49-929a-4284-a858-8a8f9802cf95

You mean it works fine for you with duo phone ?

Thank you

 

[AvayaHE]

Had a similar Problem 2 years ago

Maybe try that way :
Change Trunks
Start arrangement inc/outgoing from IMM to WNK
In DCH DATA
OVLS to NO
OVLS to NO

 

[nortavaya]

I did this changes, but the same issue,

When I call from Polycom phone to Unistim, I got busy tone, and no informations displays on TTY DCH Monitoring

 

[roc221]

no it didnt work the programming that avaya has for DUO is WRONG !!!!

 

[nortavaya]

Hi roc221,

Do you know what is the correct programming for Polycom ?

Thank you in advance

 

[roc221]

1. Start the configuration with clicking on top tab “Simple Setup”:
1) Click expand “Country” and make sure USA (default) is selected.
2) Expand Language and make sure Phone Language is set to English.
3) Expand Time Synchronization and make sure SNTP Server is pool.ntp.org and Time Zone is set to Pacific Time.
4) Expand SIP Server and make sure address is set to 10.6.10.10 and port is set to 5070.
5) Expand SIP outbound proxy and make sure address is blank and port is 0.
6) Expand SIP Line Identification and make sure the Display Name is the phone number, Address has ext number and @portseattle.org, Authentication User ID is ext number, Authentication password is the phone ext number and Label the phone in format of Site-Conference Room name format (i.e. STOC-Pike Place) and click Save.

2. Next step is to configure SIP settings (hover with cursor over Settings and click SIP in drop down):




1) Expand Local setting and make sure Local SIP Port is 5060; Calls per Line Key is 1; New SDP Type is disabled;
Live Communication server Support is disabled; Digitmap and Digitmap Timeout should listed as shown in screenshot below; Remove End-of-Dial Marker should be enabled and Digitmap Impossible match is set to 2.
2) Expand Outbound Proxy and make sure Address field is empty; Port is 0 and Transport is DNSnapt.
3) Expand Server 1 and make sure Address is 10.6.10.10 ; Port is 5070; Transport is DNSnapt; Expires 3600; Register is set to Yes; Retry Timeout (ms) is set 10; Retry Timeout Count 3; Line Seize Timeout (s) is set to 30.
4) Expand Server 2 and make sure Address field is empty; Port is 0 and Transport is DNSnapt; Expires (s) is set to 3600; Register is set to Yes; Retry Timeout (ms) is set 10; Retry Timeout Count 3; Line Seize Timeout (s) is set to 30.
5) Click Save.

3. Next step is to configure Line 1 (click on Settings and Lines in drop down menu)



Expand Identification and make sure Display name is listed as a 10 digit phone number (provided Telephony has assigned the extension number); Address should read 4 digit ext number@portseattle.org; Authentication User ID should list extension number ; Authentication password field should be same as extension number; Label should list the Site name-Conference Room name (i.e. STOC-Pike Place); Type is set to Private; Third Party Name filed should be empty; Number of keys is 1; Calls Per Line is 1; Ring Type Low Thrill (by default).
2) Expand Outbound Proxy and make sure Address field is empty; Port is 0 and Transport is DNSnapt.
3) Expand Server 1 and make sure Address is 10.6.10.10 ; Port is 5070; Transport is DNSnapt; Expires 3600; Register is set to Yes; Retry Timeout (ms) is set 10; Retry Timeout Count 3; Line Seize Timeout (s) is set to 30.
4) Expand Server 2 and make sure Address field is empty; Port is 0 and Transport is DNSnapt; Expires (s) is set to 3600; Register is set to Yes; Retry Timeout (ms) is set 10; Retry Timeout Count 3; Line Seize Timeout (s) is set to 30.
5) Expand Call Diversion Always Forward is Enabled; Always Forward To Contact is empty; If Busy, Forward is Enabled; If Busy, Forward To Contact field is empty; On No Answer, Forward is Enabled; On No Answer, Forward To Contact field is empty; No Answer Timeout (seconds) is set to 55; On Do Not Disturb, Forward is Disabled; On Do Not Disturb, Forward To Contact field is empty; Disable Forward For Shared Lines is set to Yes; Forward Specific Caller is Enabled.
6) Expand Message Center and make sure that Subscription Address field is empty; Callback Mode is set to Registration; Callback Contact field is empty.
7) Click Save.


4. Next step is to set Audio Codec Priority (as shown in next image).

 

[nortavaya]

Hi roc221,

Apology for the delay of my response

I have tried this procedure but still not working, could you just provide the digit map configuration of the Polycom, it is not define on your procedure

Before the incoming call works, only with outbound we have problem, but after applying this procedure no inbound or outbound is working

Thank you.

 

[AvayaHE]

did you restart vtrk application ?

 

[AvayaHE]

Try tssSLG logs a verry helpful tool
The logs will store in sscommon.log in /var/opt/logs or so of signaling server.

 

[nortavaya]

Thank you roc221 for the procedure, it works fine now

I appreciate your help

 

[SharkySeph]

What was the solution to this? We are having a similar issue with another phone in the PolyCom vein (Spectrlaink 8440) when we try and register via TCP as opposed to UDP.

Thanks!

 

[nortavaya]

Hi SharkySeph,

I followed the procedure of roc221, please see below:

1. Start the configuration with clicking on top tab “Simple Setup”:
1) Click expand “Country” and make sure USA (default) is selected.
2) Expand Language and make sure Phone Language is set to English.
3) Expand Time Synchronization and make sure SNTP Server is pool.ntp.org and Time Zone is set to Pacific Time.
4) Expand SIP Server and make sure address is set to 10.6.10.10 and port is set to 5070.
5) Expand SIP outbound proxy and make sure address is blank and port is 0.
6) Expand SIP Line Identification and make sure the Display Name is the phone number, Address has ext number and @portseattle.org, Authentication User ID is ext number, Authentication password is the phone ext number and Label the phone in format of Site-Conference Room name format (i.e. STOC-Pike Place) and click Save.

2. Next step is to configure SIP settings (hover with cursor over Settings and click SIP in drop down):




1) Expand Local setting and make sure Local SIP Port is 5060; Calls per Line Key is 1; New SDP Type is disabled;
Live Communication server Support is disabled; Digitmap and Digitmap Timeout should listed as shown in screenshot below; Remove End-of-Dial Marker should be enabled and Digitmap Impossible match is set to 2.
2) Expand Outbound Proxy and make sure Address field is empty; Port is 0 and Transport is DNSnapt.
3) Expand Server 1 and make sure Address is 10.6.10.10 ; Port is 5070; Transport is DNSnapt; Expires 3600; Register is set to Yes; Retry Timeout (ms) is set 10; Retry Timeout Count 3; Line Seize Timeout (s) is set to 30.
4) Expand Server 2 and make sure Address field is empty; Port is 0 and Transport is DNSnapt; Expires (s) is set to 3600; Register is set to Yes; Retry Timeout (ms) is set 10; Retry Timeout Count 3; Line Seize Timeout (s) is set to 30.
5) Click Save.

3. Next step is to configure Line 1 (click on Settings and Lines in drop down menu)



Expand Identification and make sure Display name is listed as a 10 digit phone number (provided Telephony has assigned the extension number); Address should read 4 digit ext number@portseattle.org; Authentication User ID should list extension number ; Authentication password field should be same as extension number; Label should list the Site name-Conference Room name (i.e. STOC-Pike Place); Type is set to Private; Third Party Name filed should be empty; Number of keys is 1; Calls Per Line is 1; Ring Type Low Thrill (by default).
2) Expand Outbound Proxy and make sure Address field is empty; Port is 0 and Transport is DNSnapt.
3) Expand Server 1 and make sure Address is 10.6.10.10 ; Port is 5070; Transport is DNSnapt; Expires 3600; Register is set to Yes; Retry Timeout (ms) is set 10; Retry Timeout Count 3; Line Seize Timeout (s) is set to 30.
4) Expand Server 2 and make sure Address field is empty; Port is 0 and Transport is DNSnapt; Expires (s) is set to 3600; Register is set to Yes; Retry Timeout (ms) is set 10; Retry Timeout Count 3; Line Seize Timeout (s) is set to 30.
5) Expand Call Diversion Always Forward is Enabled; Always Forward To Contact is empty; If Busy, Forward is Enabled; If Busy, Forward To Contact field is empty; On No Answer, Forward is Enabled; On No Answer, Forward To Contact field is empty; No Answer Timeout (seconds) is set to 55; On Do Not Disturb, Forward is Disabled; On Do Not Disturb, Forward To Contact field is empty; Disable Forward For Shared Lines is set to Yes; Forward Specific Caller is Enabled.
6) Expand Message Center and make sure that Subscription Address field is empty; Callback Mode is set to Registration; Callback Contact field is empty.
7) Click Save.


4. Next step is to set Audio Codec Priority (as shown in next image).

 

[SharkySeph]

Hello nortavaya - I gave that a shot and it does register, but what are you seeing if you were to connect to your signaling server and run an "slgSetShowByUID" on the device? Are they registering up TCP or UDP?

Thanks!

 

[nortavaya]

It will use the protocol what you configured on the Polycom side and NodeIP (Signaling Server)