Does anyone have ringinuse=no and autopause=yes working together in
queues.conf?
We assign members to our customer service queue from an application
based on actions the agents take on their PCs. No static agents are
defined in agents.conf and no members are specified in queues.conf. All
member interfaces are SIP with only the basics configured in sip.conf.
Even with 'ringinuse=no' configured, the Queue application continues to
send callers to busy members causing them to get paused when their SIP
device returns that it's busy.
Does the Queue application need hints for member interfaces to determine
their status?
Thanks,
James
James Fromm wrote:
No, call-limit is not being used. Do you have ringinuse=no working? Has
anyone seen it work?
Each SIP device has a very minimal config in sip.conf. Here's a show
sip peer:
* Name : 3207
Secret : <Set>
MD5Secret : <Not set>
Context : outbound
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : [EMAIL PROTECTED]
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 0
Dynamic : Yes
Callerid : "Sam" <3207>
MaxCallBR : 384 kbps
Expire : 40
Insecure : no
Nat : RFC3581
ACL : No
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 216.239.128.189 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 3207
SIP Options : (none)
Codecs : 0x8000e (gsm|ulaw|alaw|h263)
Codec Order : (ulaw:20)
Auto-Framing: No
Status : OK (14 ms)
Useragent : PolycomSoundPointIP-SPIP_650-UA/2.0.3.0131
Reg. Contact : sip:[EMAIL PROTECTED]
Watkins, Bradley wrote:
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Fromm
Sent: Thursday, January 18, 2007 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue and Interface time out
I guess I'm missing something else. 'ringinuse = no' doesn't change
anything. While on a call, the queue still sends another call and
proceeds to set the member paused after receiving 'Busy Here' back
from the SIP device.
My queues.conf is:
[general]
persistentmembers = no
[customerservice]
persistentmembers = no
musiconhold = default
reportholdtime = no
strategy = leastrecent
timeout = 20
retry = 5
wrapuptime = 30 ;allow agents 30 seconds to wrap up work
maxlen = 0 ;unlimited callers on hold
servicelevel = 60 ;calls must be answered within 60 seconds
announce-holdtime = no
autopause = yes
ringinuse = no
joinempty = yes
leavewhenempty = no
Am I missing something obvious?
What do your SIP peers look like? Are you using the call-limit feature?
- Brad
The contents of this e-mail are intended for the named addressee only.
It contains information that may be confidential. Unless you are the
named addressee or an authorized designee, you may not copy or use it,
or disclose it to anyone else. If you received it in error please
notify us immediately and then destroy it.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
