Hi,
Just did a stupid test.. listening to voice only and looks like working fine... but as soon as you read voice + screen then the system start to give few error like a synchro problem for exemple you press pause and then play and the sound dissapear....
Hi,
I'm not using citrix but just windows terminal server...
In fact It is not a really solution that i like since looks like there is several bug regarding the playback here but unfortunately i do not have much more choice.. my Logger did not contain any Board to connect the RAP unit. I have...
Hi,
I have this message on the query tool from the Remote user PC. from the server CLS if i launch the query tool there is no problem i can retrieve the calls otherwise from a remote connection i'm just not able to open it not...
David
Hi,
I have an issue making call playback trough Terminal server connection... i connect to the terminal server then to the nice unverse web page.
as soon as for example you push the pause button and start playing again then the voice dissapear and sometime the call loop on a small part of the...
Hi,
thanks regarding the autorecovery we have try but it is not really succesfull and it is not enough flexible... anyway I guess the solution is to get a stable link.. I hoep the 45 M connection will arrive soon...
I have a s8700 CM3 on 1 site and i have an ESS with g650 in another site... we have some issue with the link and sometimes it goes down unfortunately when the link is up the only solution to recover it is via the command get forced .....
any idea how this can be done automatically?
thanks a lot i will have a try as far as i understand the telephone are registering in priority on the clan of the same region but i'm not sure 100 % of that... my idea is to have few phonex and making sure they are registering on a specific clan
Hi,
I have all the clan on 1 ip range as ex 171.19.10 / 24
I want to put on the clan located is a specific place a different ip-network-region.... is that feasible even if the clan are in the same network subnet....
what kind of trouble can appears doing that ?
Hi,
I have define a SIP trunk on the s8700 adn i would like to define the asterisk PBX in front of it with the sip connection to the pbx but i have no idea about how to configure the sip.conf.... anyone already did that with an AVAYA PBX ?
DM
Hi,
I followed your suggestion but it was blcoking me.... looks like there is something blocking with the remote server.. so putting another server name then everything was fine and i was able to change and remove my trunk group.... anyway my SIP trunk is not working with asterisk it is...
Hi,
I have been creating a sig group SIP and a trunk group with ip trunk but during my test i have delete the sig wo having the trunk group removed... and now the port assign to that trunk group are blocked and i cannot change/remove/busy the trunk group the pbx tell me Error encountered, can't...
correct we are using v8.9
regarding the service pack i just know that the person who was on site installed sp3 on the CLS and nothing more regarding the logger. I have receive the doc from Nice about the spanning port and looks like ok... the only issue i have is that for a strange reason...
hi,
we put the span on the port where the traffic is passing for sure (router port)... there is no other path possible to go trough.. regardnig the SPAN port is there any specific configuration regarding this one ?
correct but i guess that if 1 CLAN is not replying and still available the telephone trying to register are not going to register until forever on the faulty CLAN because there is a load balancing between all the PBX board...
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