Displaying 20 results from an estimated 1000 matches similar to: "incominglimit stuck in app_queue"
2003 Nov 28
4
call waiting disable in sip
Hello,
is there a way to disable call waiting in sip? I`m using grandstream 101
and even when the phone is in use I hear ringing in the headset. It is
pretty annoying , is there some way to disable this? I cant find
anything like it in the grandstream docs.
Thanks
--
Anton Yurchenko<phila@dg.net.ua>
Digital Generation
2003 Jul 03
3
Using switch =>
hello,
I have a test setup with 2 asterisk servers, each having a one snom 100
via sip using it. I`m experimenting on how trunking between them would
work. I have them setup for RSA authentication which I plan to use in
the future.
So I`ve setup the keys and servers seem authenticate to each other. One
is named phila and other hurricane.
Here is what I see on phila:
-- Registered
2003 Dec 12
2
Dlink DG-104SH
Hello,
Anybody has it working with asterisk? Could you share your experience (
good/bad)
Thank you
--
Anton Yurchenko<phila@dg.net.ua>
Digital Generation
2003 Nov 24
3
strange SIP authentication/authorization behaviour
When I have an ip hardphone username setup in my sip.conf :
[109]
type=friend
username=ipphone9
secret=bla-la
host=dynamic
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
defaultip=172.20.0.139
mailbox=109 ; Mailbox for message waiting indicator
callerid=ipphone9 <109>
callgroup=1
pickupgroup=1
and this user has a wrong password then calls are denied, but
2003 Jun 27
2
Making calls from snom 100
Hello,
I`m trying to make a call from the snom 100( SIP mode) but whatever
number I dial I get a 404 error from Asterisk. Here are my configs and a
dump from "sip debug" . But if I make a call from a Zap line (see
extension 2382031), it rings the snom phone
sip.conf:
------------------------------------------------------------------------------
;
; SIP Configuration for Asterisk
2003 Oct 20
3
Call Waiting on SIP phones
Hi All,
This is the first time I'm submitting a patch, and I hope it fixes more than
it breaks. I'm putting it here, since John Todd mentioned a while ago about
the heavy load Mark and crew have at Digium (doing such good work), so I
thought all of us could test this first, and if ok submit for inclusion in
CVS later if appropriate.
This is an extension to work done earlier (sorry I
2003 Nov 17
5
Struggling with grandstream sip to asterisk
Hello.
I had grandstream working fine to FWD through my firewall.
Now I want it to talk to the asterisk server.
Did lots of reading, attempts but I keep getting registration errors even
though I can call to/from the sip phone from an analog phone on a tdm400
card.
Basically.
grandstream = 192.168.1.70
asterisk = 192.168.1.1
The error I see is ;-
-- Executing Dial("Zap/2-1",
2004 Jan 23
3
SIP register/auth with Grandstream BudgeTone-100
Hello,
I have a problem with asterisk and Grandstream BudgeTone-100.
With default configuration everything works (in anonymous mode and fixed
IP), but if Im trying to enable registering, it dos not work.
I used 'sip debug' and verbose level 10, nothing happens if I switch
telephone on (no messages about bad auth etc). As I understood, after
switching phone on at first it will try to
2003 Jul 28
2
"immediate=yes or Compleate recieved" with intcoming calls with new CVS
I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot
recieve the the calls from the zaptel interface which is a E100P with
pri signaling.
That is something with asterisk becouse rolling back to version from
06/23/03 using the new libpri and zaptel fixes the problem.
Here is an exept from the config:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension
2006 May 02
3
Sip show inuse
I have recently upgraded to 1.2.7.1 from 1.2.4.
I can no longer use "sip show inuse".
Below is the output... I know there are current calls:
redhat*CLI> sip show inuse
* User name In use Limit
* Peer name In use Limit
Does anyone have an idea why this isn't working?
Thanks,
bp
2004 Jun 23
1
Problem with incominglimit and outgoinglimit
Hi,
I seem to have a problem with chanisavail and the call limits on sip
phones(incoming and outgoing)
The problem seems to be that chanisavail when trying create to create
channels and hanging them up afterwards screw up the current usage limit on
the phones.
Example with chanisavail:
Phone A calls voicemail (usage now 1)
Phone B tries to call Phone A and uses ChanIsAvail in the dialplan.
2004 Apr 20
3
Limiting incoming SIP calls & Original CallerID on transfer
I have 2 issues which I need to resolve on our production Asterisk
server:
We are currently using Polycom IP600 VOIP phones for our office which
are capable of handling 2 calls per SIP registration. What we're finding
is when staff are on the phone, Asterisk will pass them a second call
which will show up on their display, and an audible beep is heard over
the phone (regular call waiting). I
2009 May 29
1
CAll-limit or incominglimit ?????
Good morning
How I use the described commands below to limit the number of simultaneous
calls saw voip providers that they can be effected and be received in the
trunk in the Freepbx?
I verified the commands incominglimit and call-limit as I can use asterisk
is version 1.4!
It would like to restrict for I number it to four of calls that can be used
in one trunk of a voip provider?
thanks.
2004 May 22
1
app_queue and app_groupcount
The new app_groupcount looks great for most applications but it a is a
step back for call queueing...
since app_queue calls physical interfaces and not extensions,
app_groupcont can't be used to limit the calls passed to a dynamically
added agent.
I presently use the broken sip incominglimit feature (even though it's
less than ideal as it also limits outgoing calls preventing
2003 Aug 07
1
Sip Trunk config
incominglimit is already implemented for SIP. Just specify under the
endpoint how many incoming connections are allowed.
For example,
[cisco]
type=friend
username=cisco
secret=blah
nat=yes ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms away
2004 Jul 13
0
WARNING: Deprecated incominglimit and outgoinglimit
For those that don't read every line of source code here's something I found out today...
--------
Deprecated incominglimit and outgoinglimit
Incominglimit = number of calls the local extension can originate to Asterisk.
Outgoinglimit = number of calls Asterisk will terminate to local extension.
End of Life for these commands announced**, please use setgroup and checkgroup, that will
2003 Nov 17
1
ISDN debugging and SIP dial-in issue]
(I have some problems with my mailing-list alias, I hope this
doesn't get sent twice)
On Sat, Nov 15, 2003 at 04:35:20PM +0100, Philipp von Klitzing wrote:
Thank you for your comments Philipp:
> > - with a SIP phone configured as 192.168.1.190, and with its SIP
> > server being 192.168.1.190
>
> That doesn't look right. Do you have another "SIP
2003 Dec 23
2
Asterisk + CRM
Hello,
Anyone aware of any CRM products projects that intagrete with *? Or that
integrate with any telephony products? Is there some open API for such
integration, or are they all proprietory?
Thanks
--
Anton Yurchenko<phila@dg.net.ua>
Digital Generation
2003 Nov 28
4
Mute button in Grandstream?
Hello,
Has anybody been able to get the Mute button work on grandstream? it
simply does nothing. Only Hold is avalable, which is not that good.
Thanks
--
Anton Yurchenko<phila@dg.net.ua>
Digital Generation
2004 Apr 12
3
Hunting S(n)IPs
Hi Akk,
If this has been discussed/done then apologies be-4-hand. I
did not find it in the Wiki or the Archives. Here's the
question.
We have incoming PRI lines, all on the same main number. An
attendant is supposed to handle all incoming calls. Now,
let's say I have a multi-line SIP phone. For argument's
sake (and to keep it simple) say I only have two lines.
We'll call them