SIP Sets dialing issue

[Duaneness]

CM 7.0, SMGR 7.0, SessMgr 7.0.

Two SIP test sets - one J179, one 9611G. Both have the latest SIP firmware for their model.

I finally managed to get each set registered to the Session Manager. I can call each from my H323 set on my desk. On either SIP set, if I go off-hook and dial an internal extension (x1xxxx), I get wave-off after the first digit. If I dial an external number (9-1-area code + number) it completes. If I use the "new call" soft button on the J179, I can call internal extensions. The 9611 does not have that feature.

So, obviously I've got something screwed up somewhere. I've tried using AAR as the SIP trunk, and I've tried using my actual SIP trunk, and I've tried using the SIP route pattern - but it always gives wave-off (actually it sounds like the SIT tone).

Can anyone give me a shove in the right direction?

Thanks in advance -

Duaneness

 

[bignose21]

usually you configure an OPTIM trunk (SIP TLS port 5062) SM-CM, build application sequences and assign them to the phone, at the CM end have a Route pointing to the OPTIM and stick that on the last page of the station.

 

[Duaneness]

Here's the weird thing: if I program an autodial button on the SIP set to call my (H323) extension, it works. If I manually dial my five-digit extension, it waves off after the first digit.

Further testing shows that this only happens when dialing extensions that start with a "1" or a "2". Unfortunately, all the extensions in this building start with a "1".

I'm close, but missing something.

Any thoughts?

- Duaneness

 

[ZeroZeroOne]

Here's the SIP Checklist I use. It was originally from another post in the forum so I won't take credit for creating it.

PPM Check List
Use the following checklist for troubleshooting PPM or installing items for the first time:

In System Manager
[ul][li]Under Services / Inventory / Manage Elements[/li]
[ul][li]Add CM element and synchronize[/li][/ul]
[li]Under Elements / Routing[/li]
[ul][li]Add CM SIP Entity[/li][/ul]
[ul][li]Add CM Entity Link[/li][/ul]
[ul][li]Add CM Routing Policy[/li][/ul]
[/ul]
Note: Dial Patterns are not needed for SIP phones but are required for VDNs, other extensions and server-to-server routing (Voice Mail, Skype, etc.)
[ul][li]Under Elements / Session Manager / Application Configuration[/li]
[ul][li]Add Application for CM. CM SIP Entity & CM Element added previously.[/li][/ul]
[ul][li]Add Application Sequence with CM App as the only sequence.[/li][/ul]
[li]Under Users / User Management / Manage Users[/li]
[ul][li]When adding a SIP user, Origination/Termination Sequence is set to CM App Sequence[/li][/ul]
[li]Under Elements / System Manager / Global Settings[/li]
[ul][li]Check the box next to Allow Unsecured PPM Traffic[/li][/ul]
[ul]If TLS security is required, leave this box unchecked and update certificates on the phones.[/ul]
[li]Under Elements / Routing / SIP Entities[/li]
[ul][li]Navigate to the Session Manager(s) and click Edit.[/li][/ul]
[ul][li]Verify the correct Listening Ports are set for Endpoints and the correct domain.[/li][/ul]
[/ul]
In Communication Manager
[ul]
[li]Run the command change trunk x (x= trunk group created for SM)[/li]
[ul][li]Set Numbering Format: private[/li][/ul]
[li]Run the command change route x (x= route pattern added for SM)[/li]
[ul][li]Group No = SIP trunk to SM that matches the SIP entity created in SM; FRL=0; Numbering Format=Lev0-pvt[/li][/ul]
Note: If there are multiple SIP Trunks to ASMs or BSMs, include those as appropriate for server and network redundancy.
[li]Run the command change private-numbering 0[/li]
[ul][li]Create Ext Code that matches the sip extension range[/li][/ul]
[/ul]

Adding a User in System Manager should create a station in CM. If the SIP trunk was not set manually or via a Template, you may need to update.
[ul]
[li]Run the command change station xxxx (sip phone previously created in SM)[/li]
[ul][li]On Page 6: SIP Trunk: RPxxx (where xxx is the SIP Route Pattern)[/li][/ul]
Note: Older versions of CM only allowed AAR, ARS, or one SIP Trunk. If your version of CM does not allow for the Route Pattern, use AAR instead and add AAR Analysis Entries:
[li]Run the command change aar analysis 0 (For CM v6 or if using station SIP Trunk AAR)[/li]
[ul][li]dial string created for sip extension range with route pattern created previously for SM and Call Type=lev0[/li][/ul]
[/ul]

For additional information on why private numbering is preferred, see Avaya PSN020279u.
SIP stations may register and work for outbound dialing without proper PPM. Missing feature buttons are a major indicator that PPM is not set up correctly. If problems persist, check the 46xxsettings.txt file for these settings:

SET CONFIG_SERVER_SECURE_MODE 0
SET ENABLE_PPM_SOURCED_SIPPROXYSRVR 0​