Mastered Conference Issue on IP sets

[Don]

I am running release 7.1 f540122a. I have a customer who is using the Automatic Add-on conference/Announcement feature (Master Conference) in order to page over the phones. They using a mixed bag of 4038IP's, 4018IP's, and 4037IP sets. Starting about a month ago users would complain they were not able to page using the program button on there phones. When troubleshooting I entered command of compvisu 1 0 (address of associated GD) and found compressors free and available. When doing cfnstat I find all compressors available. There are no associated errors in incvisu when issue occurs. I am able to clear the issue by executing rstcpl 1 27 (associated virtual GPA for Conferencing). Has anyone ran into this issue before and if so can you give me any insight? Are there any other traces I can do to further troubleshoot the issue?

Thanks!

[torrentula]

Please explain what happens when the user tries to page. Do they get busy signal? Or does it connect like they are paging but the audio just doesn't come across the phones?

I know when I have used the mastered conference to page before, it takes awhile for all of the phones to be added to the conference, seems like it joins them one at a time.

[Don]

Torrentula,
Sorry for the late reply. When users try to page they recieve "Network Congestion" on the display of the phone initiating the conference. The audible beep can be heard on the phones in the conference list but no verbal audio can be heard. When I do compvisu for the GD board all compressors are in service and free (when conference is not initiated). When I do listincall I can see all sets in the conference list being dialed (when conference is initiated). Initially a reset of the 1 27 fictif board fixed the issue. Now when I try to reset 1 27 it does not clear issue, however the fictif board does come back in service as well as the compressors.

Anyones insight is appreciated.

[frank]

Don,

Just curious, other conferences are working on the system ? (29 party) ?

[Don]

Frank,
There are no other conferences managed on the PCX. In recent days Alcatel has came back to me and said it may be a hardware issue with the GD/MADA3 board.

[torrentula]

Hardware could be the case. Have you tried deleting the configuration and recreating? Sometimes Alcatel can be weird like this.

[Don]

Tor,
i'll try that and see what happens, Ill also update it here.

[lbanias]

I set up an Automatic mastered (add-on)conference list with 3 IP phones and when I start the conference the phones ring once and stop, like giving beeps. If a called set picks up quickly , the other sets do not ring anymore. I added 2 external numbers in the list (abbreviated numbers) and they do the same thing. The system is a R11.1 OXE with a 1xGD3(+armada) and 2xGA3(no additional armada). Every combination compressors for Ip devices/conference(add-on/meet me) tried gives nothing. Meet-me works ok. I enabled/disabled compression at system level/intra-extra domain/set... Documentation says: Standard sets use primarily a conference circuit from their own Media Gateway. IP sets use primarily a conference circuit in their IP domain. I read somewhere that IP sets are default added to Ip domain 0, where i also created manually the IP sets range . But never saw anything there about DSP/conference... What am I missing?

[vad]

Read - system doc/ user services/ mastered conference - configuration procedures - timers (in your case timer 273)

[lbanias]

Thanks for the tip, I forgot to say that I modified this timer , from 5 sec to 15 sec (why is this default 5 sec, it can lead to missed conference almost anytime???) it helps in the way that it rings longer, BUT when the call is answered by any of the users , the other users do not ring anymore ! So, from the conference list only the master and 1 user can enter . I also added 2 analog users that do the same thing, no matter when an analog set initiates the mastered conference or when he is called from the list.