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
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
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, Thermal Wetland wrote:
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
a dhcp address. I use a linux gateway for dhcp, most devices
(3102+mythtv box, lynksys PAP2, bt100 ip phones, wireless and hosts) are all
dhcp
-Original Message-
From: Thermal Wetland [EMAIL PROTECTED]
Subj: Re: [asterisk-users] Anyone know of a pass through ATA
Date: Sat 15 Mar
Does anyone know of a Trixbox like install that has the hylafax integration
rolled in?
Looking for basic fax to email support.
Thanks,
Thermal
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To
Or any fax solution?
On Fri, Apr 4, 2008 at 11:46 AM, Thermal Wetland [EMAIL PROTECTED]
wrote:
Does anyone know of a Trixbox like install that has the hylafax
integration rolled in?
Looking for basic fax to email support.
Thanks,
Thermal
On Fri, Apr 4, 2008 at 12:55 PM, James Finstrom
[EMAIL PROTECTED] wrote:
elastix
thank you james for the recommendation, I will check it out.
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To
On Fri, Apr 4, 2008 at 4:42 PM, Jonn R Taylor [EMAIL PROTECTED]
wrote:
I made some install scripts based on centos 4 or 5 like trixbox but
without all the junk. It does have some fax setup stuff in it that I use on
our production servers that's been working for over a year. I you need any
I would like to hire someone to help us tweak our asterisk system for Snom
phones.
We would like to enable things like:
One touch recording
One touch park orbits
Presence
Please contact off-list if you will be able to help.
Thermal
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On Thu, May 8, 2008 at 5:20 AM, Brent Davidson [EMAIL PROTECTED]
wrote:
Which phones are you using and what software revision. I've had a crash
course in Snom phone lately and can probably help with at least the park
orbits.
-Brent
Brent,
We have the phones in the lab, we have 1 - 370,
We have an issue where Polycom's lose BLF functionality after a reboot. The
only way to fix it is to reboot the Polycoms.
Anyone else have this issue? We are using 1.4.18.
If I run 'sip show subscriptions' all the subscriptions come back after the
restart but the lights on the phones do not
We are still using 1.0.7 and did not see any patches for the 1.0.X branch.
Does anyone know if that branch is affected?
-Thermal
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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
Does anyone know if you can have multiple TE110P cards in one chassis?-Thermal
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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
On 5/3/06, Johann [EMAIL PROTECTED] wrote:
The wiki page doesn't mention the other fields in the file.TIMESTAMP:UNIQUEID:QUEUE:AGENT:ACTION:ARG1:ARG2:ARG3Depending on the action some of the fields may contain 'NONE' instead.You will
need to use the AGENT field to match who the connect
On 5/3/06, Joe Dennick [EMAIL PROTECTED] wrote:
On the wiki, there is a little perl script that can be used to parse the queue log and insert the data into a database.I've modified the script to use a MySQL database.With that, I have a cron job that shuts down Asterisk, parses the queue-log into
Hello,
I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that
the MWI indicators will never go off (The blinking red light and envelope in
the LCD).
I have tried to upgrade to 1.4.14 and all different SIP versions on the
Polycoms. I am now at 1.6.7
Here is the SIP Message
On Nov 28, 2007 11:26 AM, Bruce Komito [EMAIL PROTECTED] wrote:
I have seen this with Polycoms, ZIP2s and occassionally with Linksys 941s,
but only intermittently. Sometimes a powercycle will clear it and
sometimes not. We've never figured out what's going on, but we think it
is something
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 bw1/bw manually to the
Check your sip.cfg for the line:
feature.1.name=presence feature.1.enabled=1
I would imagine that you have enabled=0
That was it!
Thanks - Thermal
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We have Valet Park working well with 1.4.25. We have programmed the Polycom
softkeys to include a park button that does a blind transfer to the park
extension.
Has anyone gotten the a button to activate when a particular park orbit is
in use? It would be great if you could press the button to
I am running Asterisk V 1.4.22
Twice during the last two days the Context Switches on our box has gone from
about 7K to 80K in 2.5 hours. The load average would spike to 17, drop to
0.35 then spike again.
When connecting to the console 'core show channels' will list the channels
but not total
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
On Wed, Dec 30, 2009 at 8:27 AM, Matt Darnell mattdarn...@gmail.com wrote:
# Asterisk sip jbenable = yes|no : Enables the use of a jitterbuffer
on the receiving side of a SIP channel. (Added in Version 1.4)
# Asterisk sip jbforce = yes|no : Forces the use of a jitterbuffer on
the receive
I have an Asterisk box, 1.4.30 with a PRI.
A Mitel 3300 is connected to the Asterisk box via SIP trunking.
When a user calls from the Mitel through the Asterisk box the user can speak
but can not hear the far end.
But - when I route the Mitel user to echo() it works, send and receive. The
On Tue, Apr 13, 2010 at 7:43 PM, Prince Singh pri...@drishti-soft.comwrote:
1. Are Asterisk and Mittel in the same physical LAN.. or do they have a
router between them?
2. Do a 'rtp debug' at the Asterisk CLI to see where is the RTP data
being sent to
3. Probable issues:-
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
I have a virtual server with godaddy but can not compile DAHDI as it
complains that I do not have the correct kernel source.
The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest
version
Nothing to do
uname -a
, Jul 14, 2010 at 2:06 PM, liuxin nyliuxin...@gmail.com wrote:
Hi.
The best easy way is:
copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src
then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm
2010/7/14 Gareth Blades list-aster...@skycomuk.com
Thermal Wetland wrote:
I have a virtual
On Wed, Jul 14, 2010 at 8:09 PM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Hello
What will be your exact kernel version. Give me output uname -a command.
--
Regards,
Chandrakant Solanki
Thank you for the help! Here is the output:
[r...@ip-97-74-119-59 ~]# uname -a
Linux
On Wed, Jul 14, 2010 at 8:43 PM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Hi
Following steps to do...
1] # cd /usr/src/kernels/
2] # ln -s 2.6.18-128.2.1.el5.028stab064.7-i686 2.6.18-028stab064.7
Try this 'n let me know... Hope this will work fine...
Seems like that should
On Wed, Jul 14, 2010 at 8:43 PM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Hi
Following steps to do...
1] # cd /usr/src/kernels/
2] # ln -s 2.6.18-128.2.1.el5.028stab064.7-i686 2.6.18-028stab064.7
Try this 'n let me know... Hope this will work fine...
--
Regards,
On Wed, Jul 14, 2010 at 8:43 PM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Hi
Following steps to do...
1] # cd /usr/src/kernels/
2] # ln -s 2.6.18-128.2.1.el5.028stab064.7-i686 2.6.18-028stab064.7
Try this 'n let me know... Hope this will work fine...
--
Regards,
On Wed, Jul 14, 2010 at 11:34 PM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
The DAHDI Makefile looks for the kernel source by default in:
/lib/modules/`uname -r`/build
This is normally a symlink that points to the directory with the
(possibly partial) kernel source tree.
It seems that
On Thu, Jul 15, 2010 at 11:29 AM, Miguel Molina
mmol...@millenium.com.co wrote:
El 15/07/10 15:15, Thermal Wetland escribió:
On Wed, Jul 14, 2010 at 11:34 PM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
The DAHDI Makefile looks for the kernel source by default in:
/lib/modules/`uname
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
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