Displaying 20 results from an estimated 5000 matches similar to: "Limiting incoming SIP calls & Original CallerID on transfer"
2003 Nov 17
2
Hunt groups and SIP?
I would like to setup a hunt group, not a group ring, using sip phones.
Anyone done this with sip devices? Comments suggestions?
I have not had much luck with the outgoinglimit=1, incominglimit=1
stuff that I would need to get busy extinctions to work right, which is
why I'm asking on the list.
2004 Dec 14
3
sip_buddies mysql table
Not being an asterisk expert, but having been around
the block once or twice when it comes to data and the
like, I have made some observations based on the examples
given on voip-info.org Sip configs.
it appears there is an adjustment to be made in
the sip_buddies example table:
>>> name
Although set to 30 characters, I don't see where it is
limited in the text file. In theory,
2004 Dec 14
3
Problems with app_realtime
It seems that when setting qualify = 200 or qualify = yes in the database for
a sip friend/peer, RealTime does not update the registration status like it
should.
I also have several peers which have been offline and Asterisk still reports
them as registered, even though the registration seconds are only 200.
Asterisk Ver: CVS HEAD 12/1/2004
Layout of sip_buddies:
mysql> describe
2011 Mar 29
1
wrong from URI in options message
I recently configured a SIP peer which i must specify my fromuser as my
ten digit "DID". I send calls to this peer, but whenever Asterisk sends
an options message, the fromuser is "asterisk".
Is this a bug? Or is there some other config I must make ?
register = 2155551941:123456 at 10.0.138.226/2155551941~600
[peer](!)
type=peer
context=inbound
qualify=yes
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
2005 Oct 18
1
Queues and call waiting indication
Hi,
I'm running 1.2 beta1 in a mini call center.
I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a "beep beep" (call waiting) over and over again in Xlite audio.
An easy solution woud be the use of a "single line" user agent, like firefly, still
2007 Feb 22
1
Lastest SVN (1.4) and realtime call limit
Hello,
I am running version 1.4 with realtime support. I've set (for Snom phones
300/320/360) a call limit of 1 (incominglimit and outgoinglimit fields in the
database).
- When I used 1.4 SIP SHOW PEER show that it has a call limit of 1. The problem
was that when such a phone received a call and did attended transfer it
was left "in use" and could not receive new calls.
-
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 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.
2005 Feb 03
1
403 Forbidden when registering sip user database on backend
i am getting 403 Forbidden message from asterisk when
it try to register my user agent. i am basically
useing mysql through ODBC. i hvae checked ODBC
connecteion with
'ODBC Show' command.
------------------------------------------------------
*CLI> odbc show
Name: mysql1
DSN: asteriskdsn
Connected: yes
*CLI>
------------------------------------------------------
and user is added to
2004 Jul 29
2
Zultys Zip 4x4
Is anyone successfully using one of these with Asterisk? I cannot get the
phone to register, this message keeps coming up on the Asterisk console:
Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request:
Registration from '"000BEA801CA6" <sip:000BEA801CA6@hcs.net:5060>' failed
for '204.194.36.138'
The telephone LCD says "SIP registation
2004 Sep 21
3
Uniden uip200
I got a Uniden UIP200 and started to configure it and I am lost....
I have a tftp server setup on my * server and have the files unidencom.txt
and uniden<mac>.txt there. But it doesn't quite work yet. It registers as
a sip phone (sip show peers), but I cann't dial it and the display shows #1
disconnected all the time. It has firmware version BS4.59a in it.
I have no idea if I
2004 Jul 18
4
Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All
Total noob on the list so all help appreciated....
I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows).
I've plugged in two Cisco 7960 phones....
The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......
2004 May 25
1
Call Admission Control
Let's say you have a 256 Kbps Internet connection and you're using it for
voice calls. With mu-law (G.711), each call uses about 80 kbps, so you
really can't have more than 3 calls active at one time. Does Asterisk
support any kind of Call Admission Control where it would prevent you from
originating a call if it would exceed your Internet bandwidth? For example,
in this case, ideally,
2007 Sep 18
6
Limiting Simultaneous calls
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to have any such
restriction for internal calls.
Thanks
Jim
2005 Oct 18
2
SV: Queues and call waiting indication
Hi,
This issue has been discussed probably a million times on every asterisk forum in the world and I have the same problem too. Another problem you would have with the agents is that when they make an outgoing call they are not regarded as "busy" by asterisk and it sends more calls to the agent if it has call waiting enabled.
This behaviour is totally senseless since the whole purouse
2003 Sep 17
2
Sip call waiting
Hi folks,
As none of the SIP softphones that I tested can disable more than one
incoming call, I decided to implement it by software ;-) I'm attaching a
patch that does it.
To make it work, modify your sip.conf file and include callwaiting=[0|1]
at the general section, or for each peer that you wish to control.
Please note that I haven't tested it too much, and my source tree is
quite
2007 May 23
0
SIP.CONF: incominglimit and outgoinglimit
Hi all,
I have some peers configured in SIP.CONF file with parameters
incominglimit and outgoinglimit set up to 10. By doing that, I expect
that this peer will not be allowed to handle more than 10 incoming calls
and 10 outgoing calls at the same time.
However, since I upgraded to Asterisk 1.2.17, I started to face a
problem. Sometimes, calls to those peers are not connected. When I check
the
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
2010 May 12
2
asterisk-users Digest, Vol 70, Issue 25
Hi Vardan
I did same as you told and deleted the SIP information in Astdb and
restarted asterisk. but the result was same.
as you said there might be mistake in sip.conf so i am pasting both servers
configuration here..
1- nasir.server.com
[abc]
username=abc
type=friend
secret=mysecret
nat=yes
mailbox=12234568
incominglimit=2
outgoinglimit=2
host=dynamic
dtmfmode=rfc2833
context=payasyougo