Member On Hold

Hi, I have a problem with a queue, a member shows On Hold, the telephone isn’t On hold. There’s an option in the queue to avoid that a member enter on the status On Hold?

Operaciones has 0 calls (max unlimited) in ‘rrmemory’ strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/113 (Invalid) has taken no calls yet
SIP/112 (Not in use) has taken no calls yet
SIP/111 (On Hold) has taken no calls yet
SIP/110 (Not in use) has taken no calls yet
SIP/109 (Not in use) has taken no calls yet
SIP/108 (On Hold) has taken no calls yet
No Callers

Thank You!

The queue application thinks that the channel driver thinks the device is on hold. Either the channel driver actually thinks it is on hold, in which the queue application has no mechanism to override it, or the queue application has lost track of the device states, which is a bug in the queue application, and needs fixing properly.

Which channel technology driver? Which version of Asterisk? If SIP, does the peer use late offer SDP?

[quote=“david55”]The queue application thinks that the channel driver thinks the device is on hold. Either the channel driver actually thinks it is on hold, in which the queue application has no mechanism to override it, or the queue application has lost track of the device states, which is a bug in the queue application, and needs fixing properly.

Which channel technology driver? Which version of Asterisk? If SIP, does the peer use late offer SDP?[/quote]

Thank You for the reply.

The channel technology is SIP, and we don’t use SDP. The version of asterisk is Asterisk 1.4.9.

If it is a bug, we can still make some way to not allow the member is placed on hold? The telephone is Fanvil.

Thank you!

I don’t believe you don’t use SDP.

If it is a bug, and updating to the latest version doesn’t fix, and you do not have programming skills yourself, there is nothing you can do to fix it other than to provide detailed diagnostics.

However, I rather suspect the secret lies in the SDP you say you don’t have. Especially as you are using an obsolete version of Asterisk that definitely cannot cope with late offer SDP well. (I’m not sure if that is fixed in later versions. As PJSip is a re-implementation, it is unlikely to have the same bugs.)

Hi David,

I’m making probes, since yesterday afternoon we have not had the problem, all I did was restart the asterisk.

I don’t know nothing of SDP, i will search more about the topic.

Good Day!