Displaying 20 results from an estimated 6000 matches similar to: "Looking for a Snom expert"
2008 Feb 07
2
Snom 300 MWI
I think I have my echo problem solved, now i need to tackle the MWI. I
can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a
section from my sip.conf for my test phone:
[general]
context=internal
allowguest=no
allowoverlap=no
allowtransfer=yes
notifyhold=yes
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
pedantic=yes
vmexten=9998 at internal
;vmexten=*97
2016 Jun 07
2
Delay after Answer
Well, I thought I had the problem solved. Ported everything over to
PJSip and build RDNS records for the phones and the server, but I am
still experiencing the problem on incoming calls.
**
On 6/7/2016 1:00 PM, Faheem Muhammad wrote:
> I've faced the same issue. The issue was related to DNS, the reverse
> lookup query failure caused the delay around(7-9 seconds). The purpose
>
2008 Mar 14
3
Anyone know of a pass through ATA
Anyone know of a company that makes a pass through ATA?
By pass through I mean have an Ethernet switch built into the ATA, like most
desktop phones have.
All of the dual ethernet ATA's I have seen have WAN/LAN ports, not two LAN
ports.
I fooled around with DMZ etc...but it just doesn't work as well.
Thermal
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2008 Feb 07
4
Snom 300 Echo
We're deploying an asterisk-based phone system at all of our branch
offices in an effort to eliminate long-distance costs incurred from the
constant branch to branch calls. We're using the Snom 300's at all
offices for the desk phones and X100P cards to interface to 2 analog
lines. I'm having a problem tuning all the echo out of the system. So
far two branches are using the
2009 Sep 15
1
Detecting Transfer
Is there a way to detect if a call is a transfer in the dialplan? Here
is my issue: I have an office with 2 extensions. Under normal
circumstances any call that comes in should ring both extensions. I
accomplish this through a queue. The problem is that if the call is
answered on say extension 11 and the answerer wants to transfer the call
to the other phone, extension 10, transferring
2006 Oct 11
4
Multiple TE110P cards in one chassis
Does anyone know if you can have multiple TE110P cards in one chassis?
-Thermal
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061011/adbceeb5/attachment.htm
2016 Jun 07
3
Delay after Answer
I am having an issue with a couple of phones where they ring, but there
is a long delay after the phone is picked up before the audio starts.
My setup:
* Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
* Server is CentOS 7
* Quad core CPU with 16GB Ram
* 2 Snom 300 phones.
* NO NAT. Server and phone are on the same subnet with only a gigabit
switch between them.
* Digium
2008 Oct 09
1
Transfer/Park Question.
I've got a situation where I need to use a transfer to the parking lot
as hold, but am not going to use BLF indicators on the phone to pick up
the parked calls so I need to hear the 3-digit extension after the
transfer. I'm using Snom 300 phones and have tried setting a
programmable button to Key Event F_TRANSFER 700, which successfully does
the transfer but cuts off audio so you
2010 Apr 20
1
Put a call on hold with Manager
I would like to be able to place a call on hold via the manager interface
and be able to retrieve it.
The user can click a button in the Order entry form to put the caller on
hold when they are looking up information. It saves them from having their
hands leave the keyboard and press hold on the phone.
I don't see 'hold' & 'retrieve' commands for the manager interface.
2016 Aug 23
2
Audio cut-outs
I'm having an issue with some Snom 300s on a server running Asterisk
version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is _*NO
NAT*_ involved. Phones and server are plugged into the same network
switch, all on the same IP range. The server is running a Wildcard
AEX410 analog card with 2 FXO modules receiving incoming analog lines.
Occasionally, in the middle of a call, the
2010 Oct 08
2
Polycom getting DCHP address from wrong VLAN
Hello,
I have been tearing my hair out on this issue for 2 days, any help
would be appreciated.
We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch
There are two VLANs, 1(data) & 50(VoIP). When Polycoms are connected
to the switch with VLAN 50 hard coded in the config they grab a DHCP
address from VLAN 1, the PVID for the switch port.
The ports have membership in
2009 May 12
2
Is anyone keeping up with the versions?
We are still using 1.4 and were going to start testing with 1.6.0, but then
1.6.1 was released and now 1.6.2 is already in beta 2.
That seems like a lot of independent releases to maintain. I read about all
the regressions ans hurried dot releases, makes us nervous.
How is everyone doing their testing?
-Matt
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2006 May 02
3
Queue reporting seems broken.
I am trying to figure out which one of our agents is answering the calls.
According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log the
only time the queue_log puts the channel (agent) is during logoff & logon.
There is the connect & completeagent message, but it doesn't show which
channel (agent) answered the phone.
I can't even figure it our cross referencing the
2009 May 07
3
QoS & VPN
I've got multiple satellite office all linked back to the main office
via VPN. Each office has their own asterisk server which registers back
to the main office's Asterisk server. Each office also has a 1Mb
downstream / 384k - 768k upstream connection. The branches are using
Speex for their connections back to the main office. The issue I'm
having is that there are times that
2005 Jul 12
3
SNOM 360 and parking
OK, last showstopper that I just can't puzzle my way through - parking
calls with the snom phones. I get the two phones connected, I hit
transfer on one, the other phone goes to MOH and the first phone gives
me DT, so I dial 700 and hit the OK button. Call transferred, the SNOM
hangs up before I have a chance to hear which extension it parked to.
Is there a way to make the SNOM phones
2009 Jul 07
3
Automatic Gain Control
Is there any possibility of DAHDI supporting Automatic gain control on
TDM ports? I'm having issues at a couple of offices where calls made to
local numbers are fine but a when a calls from or goes to a large
percentage of long-distance or 1-800 numbers the person at the remote
end cannot hear the person in my office. Boosting the gains in
zapata.conf (I'm still using 1.4.21) to 8
2008 Feb 02
2
Polycom - Buddy Watch not a choice when adding Speed Dial
Hello,
On our Polycom phones we can not activate the Buddy Watch feature.
When you add or edit a contact, the list ends at "Auto Divert".....I know it
is the end of the list b/c the down arrow on the right side of the screen
disappears when I get to Auto Divert.
When I add <bw>1</bw> manually to the speed dial file it doesn't change
anything.
The buttons work well for
2008 Oct 08
1
Sip Trunking
I have several branch offices, each with their own Asterisk server
(version 1.4.22.1) handling their PBX functions. All of these offices
need to talk to each other. In sip.conf I created a peer entry for each
office with a username of branch-user and a friend entry for every
branch-user with the username being just the branch, for example:
[Office2]
username=Office1-user
host=10.10.80.253
2008 Mar 14
1
Group Listen on SIP Phone
Anyone know of a SIP phone that supports group listen?
Group listen allow you use the handset but what the far end says comes out
the speaker...it is F802 on a Norstar.
Thermal
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080313/a219002e/attachment.htm
2009 Dec 30
1
Force Jitter Buffer for SIP to SIP calls
We have a customer on a wireless connection that has very bad jitter. They
can hear people fine, but people have a very hard time hearing them. They
are connected via a SPA-2102.
It is a SIP client going to a SIP trunk.
Something like this in sip.conf [general] would be in effect for all SIP
clients:
jbenable = yes
jbmaxsize = 150
jbresyncthreshold = 1000
jbimpl = fixed
jblog = yes
I only want