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
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