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Messages - wyeee

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1
This is not happening all the time but when it happens, it is quite annoying.

Extension off 3300 calls an external number on Asterisk (using SIP Trunking).
When 486 Busy is returned (with no retry-after header), 3300 sends another INVITE (within a second, with different callid) to the same number and gets 486 again. And it goes on and on:

INVITE, 486; INVITE, 486... Until the caller hangs up. In Mitel internal trace I notice some timer is fired and that seems to kick off another call.

There must be a flag to turn this behavior off (through COS, SIP Peer Profile, ...)

Does anyone know how to do it?

Thanks

2
Ralph:

You are right. It's working now. Thanks so much.

I have DND set to No in COS already (when all incoming calls are 486ed). I don't know why DND is the default behavior of an analog phone.

William

3
Version 10.2

I programmed an analog phone off the embedded analog board (on ONSp1):
 Following forms are provisioned:
  - ONS/OPS Circuit Descriptors
      Uses all defaults (Transmission: SHORT; Flash Type: Normal; Ring frequency: National; Set Type: National; ...)
 - Analog Sets
      Uses all defaults (besides assigning my extension number. Non-Busy Ext. is set to No)
 - Station Attributes
      Assigned all default values (COS is set to 5 which is pretty much all defaults)
 - User and Device Configuration
      Assigned name/department/location

Now, I can
  - get dial tone from my analog phone
  - make outgoing calls from my analog phone to internal numbers (Caller ID OK)
  - make outgoing call to external numbers (Caller ID is verified OK) through existing SIP trunks

However, I can not call my analog phone from internal or external phones
 - On external/internal SIP phones, I get 486 BUSY from 3300
 - On internal Minet phones, I get "NO DISTB"

I don't think I need to provision Analog Trunks since I am using embedded analog board and all external trunks are SIP trunks.

* The thing is, I don't have any Analog Line License in License and Option Selection. Is that what prevented calls from being delivered to my analog phone? How come the outgoing calls worked without the analog line license?
* In COS form, is there anything special settings for analog phone?

Thanks,



4
Ralph:

You are the man. Worked.

Thanks a lot!!!

5
Mitel MiVoice Business/MCD/3300 / How to get an extn to return busy
« on: June 30, 2011, 12:28:34 PM »
I was trying to make one of my extn (8100) to return BUSY (486) to a DID call on SIPT but failed.

Here is my situation:

3 internal extns (8100, 8101, 8122), one external softphone connecting using SIPT.
I am setting up 8100 to use Call Rerouting #3 (for Day/Night1/Night2/1st Alt/2nd Alt) and CRAA and CRFA for #3 have all default values (no reroute in CRAA and all NORMAL in CRFA). That takes VM out of equation.

* 8100 calls 8101, in conversation
* 8122 calls 8100, gets BUSY (correct)
* outside sip phone calls 8100 (thru SIPT), gets 180 Ringing and Call Waiting shows up on 8100

Tried to disable call waiting on 8100 but can't get it to work (8100 is a MINET phone, I tried Force Device Busy if Any Line is in Used in COS ).

I could do a BUSY EXTENSION 8100 thru maintenance command to manually BUSY it but I still won't get 486 from outside (will get a 480 Temp Not Available)

Looks like it treats DID calls and INT calls differently but the Call Rerouting entries are provisioned the same way.

Thanks

6
Thanks a lot guys!

I had a weird thing happening to me yesterday. 2 of my phones (1 5212, 1 generic sip) has the MWI on all the time even if there is no message in the mailbox. Drove me crazy trying to check my provisioning.

What happened was I tried couple of days ago to use my last VM port for RAD and it failed terribly. Later I changed RAD to 1st port and it worked. So I returned the last port to embedded VM. And that somehow messed MWI up (even after rebooting the box).

I tried *91+extn (FAC: Message Waiting - Deactivate) from any extn and it worked. MWIs are not blinking anymore.

On the flip side, it is very interesting that any extn can dial *90 (FAC: Message Waiting - Activate)+target extn and turn target station's MWI on and drive the other guy crazy...

7
Mitel MiVoice Business/MCD/3300 / Re: MWI always on
« on: June 30, 2011, 11:10:33 AM »
I had the same thing happening to me yesterday. Not a call back situation. 2 of my phones (1 5212, 1 generic sip) has the MWI on all the time even if there is no message in the mailbox. Drove me crazy trying to check my provisioning.

What happened was I tried couple of days ago to use my last VM port for RAD and it failed terribly. Later I changed RAD to 1st port and it worked. So I returned the last port to embedded VM. And that somehow messed MWI up (even after rebooting the box).

I tried *91+extn (FAC: Message Waiting - Deactivate) from any extn and it worked. MWIs are not blinking anymore.

On the flip side, it is very interesting that any extn can dial *90 (FAC: Message Waiting - Activate)+target extn and turn target station's MWI on and drive the other guy crazy...

8
I changed RAD to port 1 and it all starts to work now. Thanks a lot!!!!!
I don't know why the system treats port 1 and port 20 differently.

Yes, I did change the ext. I restored port 20 for regular VM and looks like it needs a restart.

How do you restart the VM? Anything beside reboot/reset the box?

9
3300 running 4.2

It is confirmed that 6001 is never opened (I used a generic sip phone on 3300 to dial 6001 and there is no 200OK back).

I have 20 VM ports (I used to have all of them for VM and they worked fine). Now I am trying to use Port 20 for RAD.
All the remaining ports for embedded VM (1-19) woks fine.

OK, I'll try to change to the first port and see what happens.

Thanks

10
I tried to set up an RAD Hunt Group and use it for my ACD path.

This is what I did:
(ACD Path Directory Number: 8199; RAD Hunt Group: 6000; RAD VM Port Directory Number: 6001)

Detailed provisioning:

* COS:
   COS Number: 4;
   Comment: RAD
   COV/ONS/E&M Voice Mail Port:   No
   Recorded Announcement Device: Yes
   Recorded Announcement Device - Advanced: Yes
   Answer Plus Delay to Message Timer: 5
   Answer Plus Expected Off-hook Timer: 128
   Answer Plus Message Length Timer: 15     (Actual recorded announcement length is between 5-10 sec)
   Ringing Timer: 60
 
* VM Ports:
   Port ID: 20     Primary Directory Number: 6001
   (6001 is taken OUT OF embedded voice mail Hunt Group)

* Station Attributes:
   Number: 6001
   Intercept Number: 1 (default)
   COS(Day/Night1/Nigh2): 4
   COR(Day/Night1/Nigh2): 1 (default)

* Hunt Group:
  Hunt Group: 6000
  Hunt Group Mode: Terminal
  COS(Day/Night1/Nigh2): 4
  Priority: 64
  Hunt Group Type: RAD
  Phase Timer Ring: 1

  Member:
  Index: 1
  Number: 6001
  Presence: Present

* VM RAD Greetings
   RAD Set: 10
   Greeting 1: 1
   Greeting 2: 2
   Greeting 3: 3
   Times to Play: 1
(I've verified using the System Admin mailbox tool that the RAD number 1-3 are recorded/uploaded correctly).

* VM Greetings:
  Port ID: 20
  Greeting: RAD Set 10

* ACD Path:
  Path Directory Number: 8199
  Path Reporting Number: 251

  Path Options:
  Priority: 64
  Primary Agent Skill Group ID: 1 (ACD group 1 is working, without RAD/MOH though)
  Recording 1 Delay to Start Minutes:  empty
  Recording 1 Delay to Start Seconds: 10
  Recording 1 Directory Number: 6000
  Audio Settings:
   Audio Source: None
 
  Rest are default values

Now I tried the following
 1. Dial DID for 8199 from outside through SIPT
 2. Dial 6000 from internal phone
 3. Dial 6001 from internal phone

All I get were nothing but ringback tones. Looks like the RAD voice mail port (6001) is never opened. I adjusted my timer to avoid the DND situation (mentioned in some other posts). I've been trying this for a day and still can't get my RAD working.

Am I missing anything or did anything wrong? Or is there any System Options I need to adjust?

BTW, I don't have Analog License on my 3300 but I don't think it should matter. Also, Embedded voice mail/MLAA works fine.

Thanks a lot

11
The problem is my PSTN gateway doesn't like INVITE with no SDP. If I change to a PSTN gateway that could handle INVITE without SDP, then everything is fine.
I just thought by setting those flags, 3300 is forced to send out SDP with INVITE.
Thanks for your time. Really appreciate!

12
OK, I set the following (besides the forcing SDP settings above) in SIP Peer Profile:

 Prevent the Use of IP Address of 0.0.0.0 in SDP Messages: Yes

Now the INVITE from 3300 for xfer still doesn't have SDP. The ACK message, instead of using IP 0.0.0.0, uses the real IP (c=IN IP4 192.168.80.143), which is the IP for 3300.


13
3300 Rel 4.2. SIP Trunking.

Incoming call from PSTN gateway to 3300 ext A is transferred to ext B.
When A press the transfer key, an INVITE is sent out (without SDP). And when 200 OK is received, ACK from 3300 contains SDP of IP 0.0.0.0.

My  PSTN gateway doesn't like INVITE without SDP. And the result of the transfer is I get one way voice.

I changed following settings on 3300:

SIP Peer Profile
  Force Sending SDP in Initial Invite Message : Yes
  Force Sending SDP in Initial Invite - Early Answer: Yes

SIP Device Capabilites
  Force Sending SDP in Initial Invite Message : Yes

I thought these should force 3300 to send SDP in the INVITE message.
But it still doesn't.

Anyone sees similar things?
Thanks

14
Call from Mitel to itsp? So 200 OK is from itsp to Mitel, right?

I guess in my case the codec used between Mitel and Asterisk is Ulaw, since I recorded my MOH and AA in ULaw and I can hear them OK from my softphone.

15
You mean from 3300?

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