We have a weird thing happening with the caller id when a call is dialed
to a SIP device registered to 1.4.22. We're preparing 1.4.22 on a
development machine for switch to live.
The callerid number displayed on the SIP device (Polycom or soft phone)
is a full SIP URL, i.e. sip:[EMAIL PROTECTED]
I've discovered that the status of a SIP device doesn't get passed as
in-use when on an outbound call. Viewing the debug log the status is
always passed as 'not in use' when on the outbound call. The
sip_devicestate function doesn't appear to check the user object at all.
The devices are conf
Does anyone know what it means when the headset button on Polycom phones
is blinking? The blinking state is achieved by hitting the button twice
while on-hook. First press activates the headset circuit and takes the
phone off-hook. Second press deactivates the headset circuit, puts the
phone
Nevermind, I found it. I'll put up an SVN version in my dev environment
today.
Thanks.
James Fromm wrote:
I've reviewed the bugs reports. I didn't see anything that applied to
this. Have you? Could you point it out to me?
Olle E Johansson wrote:
21 feb 2007 kl. 15.50 sk
I've reviewed the bugs reports. I didn't see anything that applied to
this. Have you? Could you point it out to me?
Olle E Johansson wrote:
21 feb 2007 kl. 15.50 skrev James Fromm:
Anybody seen this behavior?
To determine if it's my config or a bug, could I trouble s
x27;t. The incorrect count can be cleared
up by ringing the interface for how ever many calls are incorrect.
Beware, removing call-limit will require a restart to take effect.
Thanks in advance for any help.
James Fromm wrote:
It does.
Eric "ManxPower" Wieling wrote:
Maybe Queue doesn
It does.
Eric "ManxPower" Wieling wrote:
Maybe Queue doesn't consider a SIP account that returns "BUSY" as "in
use". That would be the only case where I could see needing call-limit.
James Fromm wrote:
We do the same thing only we use ringinuse=no and
We do the same thing only we use ringinuse=no and autopause=yes for the
queue. With autopause, if the agent is busy their interface in the
queue gets paused. Setting call-limit for the SIP interface is the only
way to make ringinuse=no work.
Eric "ManxPower" Wieling wrote:
J
ringinuse option in queues.conf will work properly.
Has anyone else seen this issue? Any ideas?
Thanks,
James Fromm
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t port and changes the manager interface to not log every
command received to the debug log unless the debug option is set.
The diff can be found at http://www.omnis.com/queueendwait.diff.
Matt wrote:
Where is this patch?
On 2/16/07, *James Fromm* <[EMAIL PROTECTED] <mailto:[EMAIL PR
I patched 1.4.0 to add a command to the manager api in the queue
application to implement the end wrap-up time I was asking about. All
the command does is modify the 'lastcall' timestamp for the queue member
by subtracting the value of the queue's defined wrapup time.
Andrew Kohlsmith wrote:
Does anyone have a solution to allow an agent to selectively end his
wrap-up time? We define a wrap-up time of 60 seconds to allow our
agents to finish their notes from a call. In some cases, the full 60
seconds is not needed and our agents would like to end their wrap-up time.
Thanks,
Jay
__
Does anyone have a solution to allow an agent to selectively end his
wrap-up time? We define a wrap-up time of 60 seconds to allow our
agents to finish their notes from a call. In some cases, the full 60
seconds is not needed and our agents would like to end their wrap-up time.
Thanks,
Jay
Jim,
I too am a Teliax user. Talk to their technical support. IAX2 is NOT
preferred. They'll tell you to use SIP.
Jim Duda wrote:
Thanks for the reply Lacy.
Yes, I know that I am using IAX2 and not SIP for my connection to
teliax. IAX2 is the preferred protocol for connection to teliax.
'export MYIP' in the startup script for Asterisk.
Larry Alkoff wrote:
I was only trying to demonstrate that my special variable MYIP was
indeed in the environment of the shell. I suspect it's not in the
Asterisk process environment - why I dunno.
I'll look at that tomorrow but suspect I'll n
How do you start Asterisk? You need to make sure the environment
variable you want inside Asterisk is being exported. I use 'export
HOSTNAME' in my asterisk init script and it works like a charm.
Larry Alkoff wrote:
Thanks for your reply Ioan.
Very interesting. ${ENV(PATH)} works to displa
Olle E Johansson wrote:
26 jan 2007 kl. 16.31 skrev James Fromm:
Olle E Johansson wrote:
24 jan 2007 kl. 18.10 skrev Eric "ManxPower" Wieling:
James Fromm wrote:
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Con
Olle E Johansson wrote:
24 jan 2007 kl. 18.10 skrev Eric "ManxPower" Wieling:
James Fromm wrote:
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
interface from another SIP device and immediately hangi
ection in the SIP channel driver appears
suspect to me.
Eric "ManxPower" Wieling wrote:
James Fromm wrote:
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
interface from another SIP device and immediately ha
Our 650s are running 2.0.3b. The problem still exists for us. We see
the devices as members of our customer service queue stick on 'in-use'
in the Queue application while the device has no active SIP channel and
will accept calls. Removing 'call-limit' from the sip.conf in Asterisk
for the d
We also use Polycom IP650 phones. They are assigned to our customer
service department. Each SIP interface is a member of our customer
service Queue in Asterisk.
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
i
__
From: [EMAIL PROTECTED] on behalf of James Fromm
Sent: Fri 1/19/2007 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue and Interface time out
That worked. I don't understand what call-limit has to do with this. I
set it to 5. Why doe
____
From: [EMAIL PROTECTED] on behalf of James Fromm
Sent: Fri 1/19/2007 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue and Interface time out
That worked. I don't understand what call-limit has to do with this. I
set it to 5.
wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
James Fromm
Sent: Friday, January 19, 2007 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue and Interface time out
Does anyone have ringinuse=no and auto
ints 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
: 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: Asteri
s 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?
Thanks,
James
James Fromm wrote:
DoH! I missed th
= 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
I'm I missing something obvious?
Thanks,
James
James Fromm wrote:
DoH! I miss
DoH! I missed that ringinuse. Thanks!
Julian Lyndon-Smith wrote:
James Fromm wrote:
Hmm, the use of autopause in queues.conf introduces a new issue. When
a queue member is on a call, the queue continues to try to send calls
to the member's interface. Getting the 'Busy Here'
that autopause appears to pause the member interface
even when they're on another call. Am I missing something or is this
the expected behavior?
I didn't expect the Queue application to try member interfaces that are
busy.
Thanks,
James
James Fromm wrote:
NICE! That did th
Using Asterisk 1.4, on the console 'show application queue' mentions an
option 'i' that should "ignore call forward requests from queue members
and do nothing when they are requested." Does this work?
My assumption is that the member whose next according to the queue
strategy should get the call
NICE! That did the trick.
Thanks!
Julian Lyndon-Smith wrote:
try autopause in queues.conf
James Fromm wrote:
We are assigning interfaces directly to our customer service queue
through an application running on each agent's PC using the QueueAdd
Manager API command. No agents are de
We are assigning interfaces directly to our customer service queue
through an application running on each agent's PC using the QueueAdd
Manager API command. No agents are defined in agents.conf.
Does anyone have a solution to pause or remove an interface that doesn't
answer after a defined pe
I spent hours debugging this a few weeks ago.
The ${UNIQUEID} contains a period ("."). Mine are something like
.xx. When soxmix is executed to mix the in and out files, the
file types are not specified. This causes soxmix to attempt to
determine the file type by the filename's exten
dosent work then, then
its your configs. Also did you remember to reload asterisk ?
- Original Message - From: "James Fromm" <[EMAIL PROTECTED]>
To:
Sent: Monday, July 24, 2006 2:24 PM
Subject: [asterisk-users] Polycom_acd_functions SIP trouble
I'm trying to use t
prob. know this but in your configs it shows secret
commented out. Also it with a softphone if it dosent work then, then its
your configs. Also did you remember to reload asterisk ?
- Original Message - From: "James Fromm" <[EMAIL PROTECTED]>
To:
Sent: Monday, July 24, 2006
I'm trying to use the latest revision of Bweschke's branch from SVN for
polycom_acd_functions. Asterisk builds and runs without error but all
SIP devices can't register when specifying a secret in sip.conf. The
Polycom 601 I'm testing with and a copy of SJphone will not register.
IAX from Ide
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