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voip_request_terminated's picture
Joined: 2019/03/28
Points: 0

We have a client who uses their paging system to announce most of their calls for pickup because their employees are always in motion and they have about a half dozen phones strewn about with generic extension info (Mid-Floor, etc). They have had a couple occurrences where a call gets announced as being parked on parking lot XXX, and if the call gets picked up from any of the paged extensions before the pager hangs up the phone and ends the page, the call audio will get routed through the page channel and everyone will hear the customer speaking.

From what I was told, this problem is not 100% reproducible, so sometimes the call is not paged through all extensions even if it was picked up before the page ends. From what I have gathered about how paging works in Asterisk, a conference bridge is established with all extensions on mute (except the caller, so you can hear them). When the caller hangs up, the room is destroyed. So it seems like maybe the picked up call is being added to the conference bridge. The thing that doesn't make sense is the room seems to persist beyond the point of the caller hanging up?

Has anyone else experienced this? Did a search but couldn't find anything in the forums.


volodya's picture
Joined: 2017/01/05
Points: 260


Do you happen to have Asterisk full log file with call trace of such occurrence?

voip_request_terminated's picture
Joined: 2019/03/28
Points: 0

Well, I am awaiting another occurrence, then I can grab and dump the log.

eeman's picture
Joined: 2007/11/06
Points: 290

what phone model do you use? Ever since a lot of phones switched to supporting Multicast Paging, I would recommend that over the Page() application in asterisk.

1) it uses a shit-ton of less bandwidth over your WAN link for MTE customers. Actually it uses ZERO since multicast is all local broadcast traffic and does not route. Whereas Page() will consume 1 calls worth of bandwidth per extension added. So if you paged 100 phones your looking at 80,000Kbps of bandwidth.

2) the audio codec of multicast can be set to G722 which sounds a lot cleaner and clearer than g711 and adds no transcoding load on your MTE server.

Erik Smith
Thirdlane/Asterisk Support available