SIP and incoming call route

[tlpeter]

How does it realy work ?
I think i am on the right way (finaly after a day searching)
outgoing is going well but incoming doesn't work
i noticed (just) that the tab sip does not exist for the users, it shoud be there
BUT when you make a user shortcode it suddenly is there
when i look in monitor i see the call coming in but there is no match with the URI
does it all depend on the SIP tab at the user ?
and if yes why isn't it there from the start or by putting in the license




ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[jamie77]

The user tab is there when you set "Use User Data" for Local URI in the SIP Trunk form.

I use this for changing the o/g CLI for each user.

On ours (unsupported SIP ITSP!!!), I have had to create a second form with the DDI in the SIP trunk setup. Until I did this, I couldn't get I/C calls. I could then put an ICR for the DI as normal. Programming the URI to a user might be eough to get it there. IK would have to try it.

I think the SIP URI recived on an i/c call has to be programmed on the IPO somewhere otherwise it bins the call.

Avayaneeds to release a "this is how it should work" SIP document as we are all guessing.

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office

http://www.alliedcom.co.uk

Fooball is not a matter of life and death-It is far more important!!!!

 

[DavidCT]

Are you using SIP Registration? If so:

On your SIP URI Tab for the SIP Line you configured, what do you have for "Local URI"?

 

[tlpeter]

yes i have verification on
local URI is set on "use verificationname"

@jamie77 can you tell me how you did the ddi part
where exactly did you do it
i can't put in a ddi ar far i can see


ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[tlpeter]

oke i can call and receive calls now but with receiving a call i see 0123456789@xxx.xxx.xxx.xxx
how do i loose the ip adres on the screen of the phones ???

the problem off the oncoming call was the account
it needed to be a number instead off a name
but the provider says that it is an issue for avaya
that is the way it should work to provide a trunk with multiple ddi numbers
now it only has 1 number with multiple channels


ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[tlpeter]

@jamie77

your sip provider ,does it have multiple ddi numbers on one trunk ?
if yes then i have to try your way by adding a second form with a ddi number
that means that you need to add a form per ddi number
that also means then that for every ddi you need a license

i need more time for testing






ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[jamie77]

I only have 1 DDI at the moment. I have some Voiceflex trunks on test but the DDI's they gave me don't seem to work.

I don't think you need a license for each form you create but for each concurent call you make/recieve.

If it is needed for the system to have the DDI number programmed somewhere in the SIP programming it will be a pain in the arse. To have to program it into the Local Contact and into the ICR seems a little over the top.

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office

http://www.alliedcom.co.uk

Fooball is not a matter of life and death-It is far more important!!!!

 

[Drew2400]

Okay, so I've been experimenting with SEVERAL SIP providers. I have working: FreeWorldDialup, DIDWW, CallCentric and Sellvoip (currently not working).

For DIDWW I have 4 DIDs'. There are two entries under the trunk tab (you have to create 1 entry for each of the IP addresses DIDWW uses as this provides redudancy). Licensing is by concurrent SIP connection (NOT per trunk).

Under each of the SIP URI tabs on each trunk you need to define each of the incoming numbers as they are presented to you and assign them to an incoming and outgoing group). In my case, I have 99 as the outgoing as these are DID (i.e. incoming only). And yes, I have found that you need to create a route for each and every DDI that is presented to you and point it to a destination. This is a standard practice for PRI as well. I believe you can use the MSN/DDI configuration wizard to assist in this endeavor.

I have found that this works exceptionally well. You an also route on Incoming CallerID to route calls by CLID and DDI. And yes, I also get the number presented as 1234567890@xxx.xxx.xxx.xxx. If anyone knows of a way to defeat this, let me know. I have the voicemail of my cell phone pointed to a DID that is pointed directly to the voicemail of my office extension. So, if you call my cell and I don't answer, the message is left in my office phone. Through a call flow, if I call in from my cell phone, it let's me listen to my voicemail.

Outgoing is a horse of a different color. I've found that you really have to play with this to get it to work. Some want the number only with no +, some want the +1234567890, some what 1234567890@itsp.com, etc. Standard ARS practices fall into place for this. HOWEVER, using twinning, I can't get the CallerID to pass onto the twinned mobile handset over SIP. Anyone know of how to get this to work?

I hope this helps. I've found that once the Incoming is working, it works pretty flawlessly.

 

[buzzuser]

Can someone help me with my SIP trunks..

Using IPO 500

DSL to a voiceflex router then conected via lan port to my Lan 2

All the above is configured and seems to work ok but out going i cannot get a trunk even tho i can see the SC picking up a trunk. i just get silence then the call is dropped. then on incomming ddi rings the phone but i cannot hear the caller.

Stun server is fine. voiceflex can see the router, ddi are connecting ok all IP address are there. Invite is being received but when the phone is picked the trace from voiceflex shows the IPO 500 is not sending the correct info.. i can email over to anyone that wants to see the config. then is can be used as a template

Has anyone programmed up the IPO500 on SIP ?


thanks

Spenc
spenc@autelecom.co.uk

ACA IP - IPO - VOICE MAN

 

[tlpeter]

No voice could be a codec or a routing problem
do you have a vcm card ???

ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[buzzuser]

Yes i have a VCM 32. there is 4 ports at the front of the system. on my course work im sure VCM works in the background and i dont need to plug anything into the VCM ports?


SIP lic and VCM Lic are all valid too

 

[buzzuser]

Also there not IP Phones just 1 SIP trunk with 4 SIP ddi numbers

 

[jamie77]

How is your SIP line configured??


I have 2 SIP providers(Voiceflex and Voxalis) on our IP500.

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office

http://www.alliedcom.co.uk

Fooball is not a matter of life and death-It is far more important!!!!

 

[buzzuser]

I can email you the config file if you want. save me going over all the config.

My email is spenc@autelecom.co.uk.. i am based in Manchester

I can then reply with the config... it all looks ok set the codec from auto to g711 ulaw to see if that would make it work but no???

 

[cferrera]

We had to implement incoming call routes the following way to make it work.

For incoming number we used -1718XXXXXXX. The dash forces IPO to read DNIS from left to right. When we removed it does not work.

Hope this helps a bit.