Ringing length through SIP

[Tambo]

Hi,

We want to have 2 different lengths of ringing for 2 different call types delivering to the same agents on SIP. There is a stat within the options tab of the primary SIP server called  " agent-no-answer-timeout " if I change the value of this the phones ring for the lentgh of time I set but this happens for both call types.

Does anyone know if what i'm asking is possible?

cheers

Tambo

[René]

Hi Tambo,

SIP Server supports controlling of no-answer-timeout from routing strategy via parameters stored in the Extensions collection. You have to attach the key NO_ANSWER_TIMEOUT with required timeout value before call is routed to an agent.

More details can be found in SIP Server Deployment Guide.

R.

[Tambo]

Thanks Rene,

I have done this with a NULL value so that it will pick up the value set in the strategy but it is not referencing the strategy and picking up the agent-no-answer-timeout value in the SIP server.

I have been through the SIP server deployment guide but cant see where i'm going wrong. The function box in the strategy is set up with "v_WaitTime" and set to 7 but in this strategy i want it to be 15. 

Tambo

[Tambo]

Rene,

If i put "null" values against..........
no-answer-timeout
agent-no-answer-timeout
extn-no-answer-timeout
posn-no-answer-timeout
which are all in the SIP server

would this then allow the strategy to rule over the timeout value???

cheers

Tambo

[René]

Hi Tambo,

Yes, it should as values provided in Extensions collection have preference over these configured on application level. However, you shall provide all required keys (see SIP Server documentation for the list) to make it working properly.

R.

[Tambo]

Hi Rene,

thanks for this I have taken the values out and tested and it does work via the strategy.

cheers

Tambo

[Tambo]

spoke too soon..........there are 2 settings in the SIP server that seem to be controlling the timer  after-routing-timeout (this we use as the calls bounce from 1 desk to another) and recall-no-timeout (SIP guide says that we should not alter these values !!!!

any ideas on getting this to work??

cheers

Tambo

[René]

Hi Tambo,

I'm not sure I do understand what kind of issue you having with mentioned options... Could you explain it more in details?

R.

[Tambo]

Hi Rene,

We have 2 strategies which route calls to the same agents. These calls go to our retail agents within shops. 1 strategy only routes to an individual store (each with their own numbers) and the other strategy routes around the stores virtually.

We  want to make the individual stores calls ring for longer than the virtual calls.

The settings in the strategy for v_Waitime are not controlling the ringing length.

The settings in the SIP server as above do control this but does this to both call types.

I just want the strategies to contrrol the ringing length not the SIP server.

sorry I dont know if this makes more sense 

Tambo

[bublepaw]

Hi Tabo,

What is option value for divert-on-ringing?. No answer supervision only works when this is set to true. I'am assuming that per Rene suggestion You did put all Extensions NO_ANSWER_TIMEOUT, NO_ANSWER_ACTION and NO_ANSWER_OVERFLOW before sending call to an agent?. In You log files do You see entry which tells why SIPS redirected call. In SIPS there are multiple timeouts and maybe in Your case some other timeout is causing calls to be redirected instead of NAS timer. Usually log files can tell You which timer causes redirection.

Pawel

[Tambo]

Hi Pawel,

yes done the change to the settings, haven't done the logs yet (DOH!) I think the problem might lie with the strategy, this may take a lot longer than first thought. Any ideas on what block will control the ringing length as the one I thought it was only controls the wait time to route the call ??

Tambo

[René]

Hi Tambo,

It seems to me you mix several things together...

1/ URS doesn't provide any functionality allowing you to control length of ringing. That's controlled by SIP Server - check the 'sip-invite-timeout' option (default value is 32 seconds). It should be longer that configured RONA timeout otherwise RONA won't work.
2/ You shall configure parameter as suggested by Pawel to make RONA feature working correctly

When checking logs you should see following steps (call flow):

1/ Target agent is found by URS strategy - RequestRouteCall is sent to SIP Server
2/ SIP Server sends INVITE message to agent's extension (INVITE timer is started)
3/ Received 100 Trying (optional) and 180 Ringing from agent's extension (INVITE timer is stopped)
4/ Events EventRinging (SIP Server) and EventRouteUsed (URS) are generated
5/ RONA timer is started
6a/ Call is answered by an agent
6b/ RONA timer expires - call is redirected to configured RONA destination

R.

[Tambo]

Rene,

many thanks for this. Looks like I was going down the wrong road on this one.

So if the SIP server controls the ringing length, I would be right in saying that there is no way of making the agents phones ring for different lengths of time for the 2 different call types?

Tambo

[René]

Hi Tambo,

[quote]So if the SIP server controls the ringing length, I would be right in saying that there is no way of making the agents phones ring for different lengths of time for the 2 different call types?[/quote]
No, there is a way. The option 'sip-invite-timeout' says how long SIP Server waits for response to INVITE message sent out. I think this timer is stopped when SIP Server receives response like 100 Trying, 180 Rining etc. I was not accurate in my previous post in this. Sorry.

So you can still control ringing time using No-Answer supervision functionality. Just be sure that option 'diver-on-ringing' is set to to 'true'.

R.

[Tambo]

Thanks Rene,

So that means that i can have 1 call type ringing for say 10 secs and another ringing for 20 secs ?!? because when I use the No-Answer function it changes both call types to the same length.

cheers

Tambo