[Call Supervision]

Hi:

 

It is possible to limit the call supervision so who has the permissions don't can hear the conversation of specyfic extensions?

 

For example a second level manager has the permissions but He can´t hear the conversations of the General Manager's extension?

 

Thanks and best regards

 

Juan Acevedo

SE HABLA ESPAÑOL

[Bad function of Agent group and hunt group]

Hi:

 

Some days ago I have post to the PBXnSiP group with the following problem

 

I have defined one agent group with redirection timeout and

redirection target, also i have defined the gap between announcements.

 

I call the group number and the call rings at group members and

anybody can catch teh call, but:

 

1. When nobody of the members answers the call, the announcements are

not played, but reachs the redirection timeout and goes to the

redirection target.

 

2. If all the members are busy, the annoucements are played but when

the redirection timeout is accomplished, does not goes to the

redirection target and the announcements are played for ever and ever.

 

Anybody has some answer for that?

 

I have received the following answer:

 

The redirection timeout affects only calls that are in the "ringing" state already. Callers stay in the waiting state until they are old and grey. The only way out is a key that you may define. I guess we need to address this in 2.1.

 

 

I have replayed:

 

According with the previous answer , I have switched to handle hunt groups.

 

I have defined the following scheme:

 

group 300:

members ext 160 161, duration 30 secs.

finel stage: 200

 

200: AA with a waiting message "please wait" and redirect number 301 timeout 0 to reach 301 after the message is played.

 

301: another group

members: 160 161,duration 30 secs

final stage: 201

201: AA with a bye message and hungup timeout 7 secs so when the message ends, must to hung up.

 

The result is:

 

When I dial 300, 161 y 162 rings, I dont answer and 30 secs later the AA 200 plays the mesage and after 30 secs DOES NOT PLAY THE AA 201 and the call is hungup.

 

It means that the 301 group does not access the AA201 when it is called form the secuence 300-200-301-201. BUT

If I call the 301 directly, the AA 201 is called, the same if I call from the AA200.

 

Any body has any explication for this behavior?

 

By now I have not answer to my last post

 

Thanks and Best Regards

 

Juan Acevedo

Medellin, Colombia

[Delay in audio for agent group]

Hi:

 

I Have the same problem, and I dont have bandwith limitations

 

Thanks

Juan Acevedo

Medellin, Colombia

 

Hello

 

We have some problems with a delay in the audio when there is more then 4 phones ringing in the same time on the agent group.

when one of the agents take the call then is there a delay in a couble of secunds before the caller can hear the agent speek.

 

there is now problems when using a normal huntgroup.

can somebody help me ?

 

Kasper

Denmark.

[Answering machine functionality]

I received a request today that would be great to have in the product. If someone calls your extension and you have call screening set you can send it to voice mail and listen to what they person is saying and then hit a key and jump into the call. Much like we can do with a traditional answering machine at home. Does anyone else think this would be appealing to have in the product?

[To connect two pbxnsip systems together]

Hi

My name is Juan Acevedo and I m a new member for today.

 

First I have nice results with PBXnSIP, for example I have one instalation handling 120.000 calls per month, Is terrific.

 

My point is: I need to connect PBXnSIP systems working togheter, I have defined in each system a SIP registration Trunk toward the other system and one extension number to accept the registration of the other system. I call these trunk "cross trunk"

 

The systems get register each one in the other one, and I Have defined dial plans in each in order to make calls between the systems, the calls goes fine from one system to another, including the extension number of the calling party, using the $u option in the explicit remot party ID. This features carries the extension number but it does not carry the extension name originates the call. This is the first topic I need to solve.

 

The second one is more important:

 

I have just one SIP Gateway to PSTN, registered to one of the systems, when a call incomes is drived to AA of his own system and from there I need that the caller can dial directly to any extension of the other system thet does not have SIP gateway to PSTN. The caller dials the extension number wich acoording with the dial plan must drive the call to the other system, at log I can see the system recieves the DTMF code to get the extension but the call remains in silence and PBXnSIP does not make nothing with this dialed extension number.

 

At the "cross trunk" I have defined in the system receives the call I have set "visible in all dial plans but does not make any diference.

 

I will appreciate all the suggestions can make me.

 

Best Regards

Juan Acevedo

Medellin, Colombia

 

P.D: If anybody want to email me in spanish, my email is: consultorit@umi.com.co, I fact I will be happy in to exchange experiences with another PBXnSIPers Spanish speakers.