In my instance I'm using m0n0wall, but this is a hardware-neutral
question.
Sometimes, yes and no. The trick in Monowall I founds is to use the auto
add in Monowall to create the rules. If you manually create the rule, she
don' work. Why? Dunno. Using 1.11 built on Thu Nov 11 23:02:41 CET
I just needed to test a dialup modem connection (don't ask) and I had a
modem connected to a TDM card (FXS port) which then dialled out via a E1
PRI on a TE4xxp card.
See my log below:
atdt0198xx
CONNECT 36000 V42bis
** Dial IP **
Username:
Password:
Entering PPP Session.
IP
On 7/11/05, Gonzalo Servat [EMAIL PROTECTED] wrote:
On 7/12/05, Eric Bullen [EMAIL PROTECTED] wrote: I hope someone can offer me some help with this. Basically, the current CVS version of Zaptel will not compile under Fedora Core 4. I have closely
followed the directions in
On Tue, Jul 12, 2005 at 11:04:16AM +1000, Gonzalo Servat wrote:
On 7/12/05, Eric Bullen [EMAIL PROTECTED] wrote:
I hope someone can offer me some help with this. Basically, the current CVS
version of Zaptel will not compile under Fedora Core 4. I have closely
followed the directions in
Take a look at http://www.snom.com/white_papers.html,
http://www.snom.com/whitepapers/FAQ-04-03-26-v3_4-sf.pdf and check out
DHCP option 66 and 67.
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Colin Anderson
Sent: Monday, July 11, 2005
After IEEE finally released 802.3af snom supports all three modes in the
320/360 models:
http://www.snom.com/whitepapers/faq-05-03-16-da.pdf (snom 320 = snom
360).
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
chris gamble
Sent: Tuesday,
The new grandstream GXP-2000 works quite nice with the standard 802.3af
David
Christian Stredicke wrote:
After IEEE finally released 802.3af snom supports all three modes in the
320/360 models:
http://www.snom.com/whitepapers/faq-05-03-16-da.pdf (snom 320 = snom
360).
CS
Marc Fishman ha scritto:
I appreciate the response but that's what isn't working. I have tried v5.3
and v3.0 with the same result. I suspect the firmware version (P003AM30) is
I know it's hard to find out infos at the cisco site.
Maybe you can open a TAC case
Sergio
I am not aware what I have done wrong, but the result is a query of:
*Database error:* Invalid SQL: SELECT * FROM WHERE
UNIX_TIMESTAMP(calldate) = UNIX_TIMESTAMP('2005-07-01') ORDER BY
calldate DESC LIMIT 0,25
*MySQL Error*: 1064 (You have an error in your SQL syntax; check the
manual that
Dear All,
I was trying to use Realtime Asterisk option for Sip users and peers
+ Heartbeat + Mysql Replication in order to make a failover system, so
that if Ast1 went down for any reason, Ast2 server will have the same data
that Ast1 used in the Mysql database and don't need to make the phones
On 10 Jul 2005 at 22:01, Armin Schindler wrote:
On Sun, 10 Jul 2005, Clive wrote:
Hi all
I am wondering if anyone has had a similar trouble to this:
The timeout arguments in the dial command does not work. The caller
does not get disconnected when the timeout reaches
exactly, only high-end cisco 7970/71 are 802.3af compliant,
other models (7905-7960) using proprietary PoE detection and you will be
out of luck if you use non-ci$co poe equipment,
as I know, powerdsine midspans (because have cisco detection support)
can power cisco 7912 directly (without
Hi guys,
I currently have a sip proxy server (sip express router) which
registers the sip phones. I need to add voice mail capability, i.e.
sip express router will forward all incoming calls to Asterisk if the
user does not pick up the call in 15 seconds.
The voicemail recording stops when the
Dear All,
I was trying to use Realtime Asterisk option for Sip users and peers +
Heartbeat + Mysql Replication in order to make a failover system, so
that if Ast1 went down for any reason, Ast2 server will have the same
data that Ast1 used in the Mysql database and don't need to make the
phones
Hello,
I have been noticing the following behaviour with the monitor command..
Normally it records to the default location and then uses soxmix to
create the correct wav file.
But for some reason sometimes it doesn't use
/var/spool/asterisk/monitor/.. but //var/spool/asterisk/monitor/..
In article [EMAIL PROTECTED],
Mark Edwards [EMAIL PROTECTED] wrote:
Hi Tony
I am having a similar issue to you - from the 'other' direction in that
when I connect to * via IAX2 the DTMF is being ignored. I am running HEAD at
the moment.
(and for the benefit of another subscriber so
ok what softphone i should use to fit windows and linux supporting
iax,thanks in advance.
_
FREE pop-up blocking with the new MSN Toolbar get it now!
http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/
Hello
I am planning to build a small PBX using
TDM22B.
We have a Siemens Hipath 3750 in operation
already.
When I manage to complete my PBX using TDM22B
I would ofcourse like to be able to connect my Asterisks PBX
with the Siemens Hipath 3750 PBX.
Will there be any issues regarding my
In article [EMAIL PROTECTED],
Ronald_Wiplinger [EMAIL PROTECTED] wrote:
We come into this section of the dialplan:
exten = 8867033,1,Wait(1)
exten = 8867033,n,SayUnixTime
exten = 8867033,n,NoOp(If you know the extension ...)
exten = 8867033,n,Dial(${PHONE_6003})
The
Hello!
Can you please post your CGI script ?
Thanks
M.G.
- Original Message -
From: Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Sunday, April 24, 2005 1:30 AM
Hi!
I have a problem compiling the res_config_mysql.so after successfully
compiling and installing asterisk with the bristuff package/patches. I
get lots of compiler errors.
When inserting a previously compiled res_config_mysql.so into the
bristuff-patched Asterisk, I get an error on startup and
On Tue, Jul 12, 2005 at 09:52:09AM +0200, Kristof Hardy wrote:
Hello,
I have been noticing the following behaviour with the monitor command..
Normally it records to the default location and then uses soxmix to
create the correct wav file.
But for some reason sometimes it doesn't use
Hiya!
I have just moved up to asterisk 1.0.9, and also to chan_capi-cm-0.5.3.
Since the upgrade, the ${DNID} variable seems not to be set anymore.
I made the updates to modules.conf (and in the globals sections). When
in Debug, CALLERID is correct - but DNID is NULL.
any ideas?
Álainn
Hi Wasim,
Check out the x-lite softphone
http://www.xten.com/
As for linux check this page there are two softphones type available
:
http://www.iptel.org/products
Thx
MAG
wassim Darwish wrote:
ok what softphone i should use to fit windows and
linux supporting
iax,thanks in advance.
On Tue, Jul 12, 2005 at 08:31:07AM +, wassim Darwish wrote:
ok what softphone i should use to fit windows and linux supporting
iax,thanks in advance.
Depends on what you want from it.
e.g: iaxcomm is free and availble for both those platforms. OTOH, the
user interface, well, leaves some
Good
morning
on our Test Machine
based on RedHat 9 on a Pentium 4 SMP we are experiencing a 99% CPU load for
asterisk with the CVS of tonight
does anyone noticed
that
best
regards
Thierry
___
Asterisk-Users mailing list
Hi,
I'm using Bristuff 0.2.0-RC7k and asterisk 1.0.6 and I'm facing
something similar...
Nearly all my monitor files are in 2 parts, soxmix doesn't compile them
into one file. But I don't think soxmix is to blame because when I run
it from the command line, everything is ok... the problem
Those 2 softphones below only do SIP, no IAX.
Zoa,
--
www.asteriskguru.com
Mohamed A. Gombolaty wrote:
Hi Wasim,
Check out the x-lite softphone
http://www.xten.com/
As for linux check this page there are two softphones type available :
http://www.iptel.org/products
Thx
MAG
wassim
Not sure this applies, but a few months ago some of us were having
problems with an itsp (livevoip.com) and incoming iax calls that
hit the * ivr. dtmf was essentially not being passed. I opened a
bug, however Mark quickly closed it with a note that's the way
iax works; have the itsp fix their
Kevin P. Fleming wrote:
chris gamble wrote:
Whats a good high quality ip phone that uses IEEE power over ethernet --
or is there a problem with IEEE power over ethernet??
Polycom IP301/IP501/IP600 all support IEEE 802.3af right out of the box.
That's actually not correct. The IP600
Version 0.122 - 12 July 2005
* Call charges are now shown on the Calls page
* IPSwitchBoard will check for a live connection every minute and reconnect
if the connection is lost for some reason (asterisk restart etc.)
* Bug fixes
FREE Download: http://ipswitchboard.thorben.dk
Hi,
I'm using firefly for windows (only!!) and it seems to work well...has
sip and iax support, many account (useful for testing on many asterisk
pbx).
Giorgio.
Zoa wrote:
Those 2 softphones below only do SIP, no IAX.
Zoa,
--
www.asteriskguru.com
Mohamed A. Gombolaty wrote:
Hi
Dear All,
I have been running an Asterisk 0.7.1 (patched with various agent
applications) server for almost 2 years.
We have a data center in the USA and a call center in the UK. All calls
are routed to a group of central call queues in the USA. Agents from the
data center, call center and from
Tzafrir Cohen wrote:
Here is some logging:
monitor executing ( nice -n 19 soxmix
//var/spool/asterisk/monitor/SIP-242-027e_1-in.wav
//var/spool/asterisk/monitor/SIP-242-027e_1-out.wav
//var/spool/asterisk/monitor/SIP-242-027e_1.wav rm -f
//var/spool/asterisk/monitor/SIP-242-027e_1-* )
Hello, friends.
I need to block collect calls on my PBX.
I was able to find information on Google regarding ISDN ZAP channels, but not
ISDN CAPI channels which is my case.
Since there's no information from the Telco that the call is going to be charged
by the callee, if a particular call is
Take a look at the advanced page. At the bottom is an option. Set it to
update automatically, then the phone will not wait for user interaction on
boot-up when a new firmware is available.
Regards
Nils Ohlmeier
On Monday 11 July 2005 23:27, Colin Anderson wrote:
Anyone know how I can push a
In article [EMAIL PROTECTED],
Rich Adamson [EMAIL PROTECTED] wrote:
Not sure this applies, but a few months ago some of us were having
problems with an itsp (livevoip.com) and incoming iax calls that
hit the * ivr. dtmf was essentially not being passed. I opened a
bug, however Mark quickly
On Tue, 12 Jul 2005, Dhennys Pestana wrote:
I need to block collect calls on my PBX.
I was able to find information on Google regarding ISDN ZAP channels, but not
ISDN CAPI channels which is my case.
Since there's no information from the Telco that the call is going to be
charged
by the
Hello, i've googled and can't find a definite answer, so here goes:
I have purchased the Digium TE100P, and am setting up the connection,
however the
telco i'm supposed to work with does not support PRI/ISDN E1
connections. They only
support E1/R2 lines. Is there a way i can make the TE100P
Dhennys,
I would expect that the ISDN collect call would have some kind of
notification about the charge.
In E1/R2, the Telebras standard in fact DOES have this notification
defined, from what I remember, the problem was that many of the CO
switches would not support it, that is why the
Hello Guys...
I'm looking for someone to help me out with an [EMAIL PROTECTED] installation...
I've managed to get it working to the point that extensions can talk to each
other, but not incoming calls or outgoing calls...
I need to get it configured with Broadvoice...
I'm willing to pay
On Mon, Jul 11, 2005 at 01:16:08PM -0500, Patrick Friedel wrote:
I'm rolling out an installation with snom 360s in the near future.
Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a
snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002. I
have the 360's
It would be better if you could give us some more details about your
configuration so that someone could help.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Tuesday, July 12, 2005 3:12 PM
To: asterisk-users@lists.digium.com
which the best softphone that works with window and
linux supporting IAX2 ,thanks in advance.
__
Yahoo! Mail
Stay connected, organized, and protected. Take the tour:
http://tour.mail.yahoo.com/mailtour.html
which is the best softphone that works with window and
linux supporting IAX2 ,thanks in advance.
Sell on Yahoo! Auctions no fees. Bid on great items.
http://auctions.yahoo.com/
jonny hashem wrote:
which the best softphone that works with window and
linux supporting IAX2 ,thanks in advance.
None.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
hello
i am trying to develop perl application for asterisk
with radius accounting how can i debug that weather
callback is working when call is stoped.
how can i check this
syslog('info', 'hello Asterisk!');
thanks
Kamran
Of course. Note that I have no idea what glaw is but
someone on some board
shttps://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gif
https://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gifuggested
it as a resolution to a similiar problem so I put it in.
The
Kevin P. Fleming wrote:
Matthew Boehm wrote:
Can't it be changed so that if Server A has stored an unknown address
for
phone B that if it needs to contact B again it should look up in the
database to try and contact it instead of just giving up? Perhaps
rtagressive option? Contact only, not
I'd be interested to understand where you read about the 'sequence numbers' issue.
This sounds like it might relate to the problem I am experiencing.
It's either that, or the DTMF is coming inband over the IAX channel...
;-)
Mark
On 7/12/05, Tony Mountifield [EMAIL PROTECTED] wrote:
In
Hi,
OptiPoint 4x0/600 supports IEEE 802.3af.
--- chris gamble [EMAIL PROTECTED] wrote:
I am looking at phones for my asterisk system and
seem to have a problem.
The only Power over Ethernet phones I can find that
support the IEEE
standard are 3com. Cisco uses its own proprietary (
and
I get greeted, put in a queue, given my position, the call goes through
to my soft phone, and I accept the call, press #... I then get a message
telling me that the system saying transfer? I see nothing on the CLI
except the usual waiting for '#' to acknowledge
Send the complete
Hello,
I am looking to replace my company's Avaya Merlin Magix system with
an Asterisk based PBX when our current lease is up. I had a meeting
with upper management yesterday, and they would like some assurance that
other companies are running Asterisk with success. We are a relatively
We initially tried going with Snom but had a high failure rate. Since
then we have been Polycom and they seem to work great.
We did just have a customer demand a phone with a sidecar and bought
a Snom 220 with 2 of them. The additional sidecars require a seperate
power supply which did not
Hi,
we really need the feature Call Pickup with CID info
http://www.voip-info.org/wiki-Asterisk+bounty+Call+Pickup+with+CID+info.+-+SIP
in the current Asterisk release because we have a newer TE405P card
which needs 1.0.8 or newer to work.
The call pickup patch only works for 1.0.7. Who is
Rana Dutt wrote:
Please, please add this option. If you send me a patch, I will gladly
volunteer to test it thoroughly.
I did not volunteer to write it, only commented that it would be an
acceptable option to add. My project list is already quite long :-)
Having both MWI working and
Eric Wieling aka ManxPower wrote:
That's actually not correct. The IP600 supports PoE out of the box.
The IP 30x and 50x support PoE with a special cable from Polycom.
Bummer...I thought the built-in PoE chip was one of the few upgrades in
the 300-301 and 500-501 paths... too bad, would
In article [EMAIL PROTECTED],
Mark Edwards [EMAIL PROTECTED] wrote:
I'd be interested to understand where you read about the 'sequence numbers'
issue.
This sounds like it might relate to the problem I am experiencing.
It's either that, or the DTMF is coming inband over the IAX channel...
Yes I have experienced the same on my test machine.
It has been like this for 3 weeks of CVS Head.
Someone must havea look at that, I think is
the SIP channel.
- Original Message -
From:
Thierry Wehr
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 12, 2005
The statements about Cisco and PoE aren't strictly correct.
Cisco 7940 and 7960 phones without the G (global) suffix used Cisco PoE
and had their keys labled in local languages. The 7940G and 7960G Global
phones are IEEE 802.3af PoE and have the keys engraved with icons and stick
over labels.
Title: NAT=YES
Good morning
Does anyone have experience with NAT=YES? I have the following configuration and am a bit confused as to why the Asterisk server initially sends out RTP to the remote host private IP and then switches to the public IP.
Configuration Info:
I have all users in
Hi,
new features:
RFC 3261 compliance (no TCP)
RFC 3264 compliance
RFC 3311 Compliance (display updates only, no media)
Remote-Party-ID for display updatesA Remote-Party-ID header received in
an INVITE or 200
OK will now update the display of the phone to accurately reflect the
I can help you! I can be reach at this email [EMAIL PROTECTED] or via IM
Yahoo jraborg, no problem, what kind of FXS or FXO are you using?
JR
It would be better if you could give us some more details about your
configuration so that someone could help.
-Original Message-
From:
Mohamed A. Gombolaty wrote:
Dear All,
I was trying to use Realtime Asterisk option for Sip users and peers +
Heartbeat + Mysql Replication in order to make a failover system, so
that if Ast1 went down for any reason, Ast2 server will have the same
data that Ast1 used in the Mysql database
What king of signaling your telco support?
try on /etc/zaptel.conf
span=1,0,0,ccs,hdb3
bchan=1-15
dchan=16
I have it working with :
switchtype = euroisdn
signalling = pri_cpe
Cheers.
JR
Hello, i've googled and can't find a definite answer, so here goes:
I have purchased the Digium TE100P, and
Michael J. Tubby B.Sc (Hons) G8TIC wrote:
Cisco 7940 and 7960 phones without the G (global) suffix used Cisco
PoE and had their keys labled in local languages. The 7940G and 7960G
Global phones are IEEE 802.3af PoE and have the keys engraved with
icons and stick over labels.
Are you sure
Christoph wrote:
Hi!
I have a problem compiling the res_config_mysql.so after successfully
compiling and installing asterisk with the bristuff package/patches. I
get lots of compiler errors.
When inserting a previously compiled res_config_mysql.so into the
bristuff-patched Asterisk, I get an
Kevin P. Fleming wrote:
Rana Dutt wrote:
Please, please add this option. If you send me a patch, I will gladly
volunteer to test it thoroughly.
I did not volunteer to write it, only commented that it would be an
acceptable option to add. My project list is already quite long :-)
Having
First off kill the Glaw. It doesn't exist.
Then try your call. But also why are you sending the line congestion
when you first start to make a call. That's normally used as a closure.
But from what I can see about the only thing wrong is the GLAW. Kill
that and you should be good to go.
Thanks for replying. Frustrating, didn't work. Set it to update
automatically, and made an HTML page consisting of:
html
pre
bootloader:
firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin
/pre
/html
Added the HTML page to a webserver I control, and added it's URL to the
Setting URL
currently we are able to use our USA sip phone to conenct into the E1 box, but still unable to dial out to chinese phone numbers. They said from their ISDN switch console, it shows D channel not connected to the voip server yet.here si the sip debug msg, we got a Message type: DISCONNECT (69) and
On Tue, 2005-07-12 at 09:42 -0500, Matthew Boehm wrote:
How do you expect me to possibly fix my module if you don't supply any
compile errors? I don't use BRI so you will need to provide me alot of
info. Which bri package did you install? Perhaps I can install that in a
tmp dir and see what
We sell an 802.3AF PoE Injector that includes a standard RJ45 patch
cable and the reverse polarity RJ45 patch cables for use with Cisco PoE
Endpoints. Cost is $29.95/ea quantity discounts are available.
http://www.voipsupply.com/product_info.php?manufacturers_id=22products_id=570
Cory Andrews
Sergio Chersovani wrote:
I know it's hard to find out infos at the cisco site.
Maybe you can open a TAC case
Sergio
I did find this info:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20cisco%2079xx
comments_threshold=0comments_offset=0comments_sort_mode=commentDate_desc
Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf
Mark
Klint, Peter wrote:
Good morning
Does anyone have experience with NAT=YES? I have the following
configuration and am a bit confused as to why the Asterisk server
initially sends out RTP to the remote host private IP
Thanks for the response.
I added the glaw when it wasn't working so just removing
it won't resolve the issue.
I have the congestion entries in there to prevent dialing
to
certain types of paid lines and mobile phones in The
Netherlands. I will route mobile phone calls to another
provider at
The problem with updating the firmware via settings is, that the information
about the firmware are not allow in the first settings file.
Instead the settings file which you entered in Setting URL needs to have the
following link:
firmware_status:
Hello
I am planning to build a small PBX using
TDM22B.
We have a Siemens Hipath 3750 in operation
already.
When I manage to complete my PBX using TDM22B
I would ofcourse like to be able to connect my Asterisks PBX
with the Siemens Hipath 3750 PBX.
Will there be any issues regarding my
Uniden, Polycom, to name 2 more.
On 7/12/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED] wrote:
The statements about Cisco and PoE aren't strictly correct.
Cisco 7940 and 7960 phones without the G (global) suffix used Cisco PoE
and had their keys labled in local languages. The 7940G
FYI, there is no such thing as reinvite. Someone started using
that in postings a long time ago and a few people keep posting
it, but it doesn't exist.
(Check /usr/src/astersik/configs/sip.conf.sample)
Add canreinvite=no and reinvite=no to the relevant stanza in
Christoph wrote:
On Tue, 2005-07-12 at 09:42 -0500, Matthew Boehm wrote:
How do you expect me to possibly fix my module if you don't supply any
compile errors? I don't use BRI so you will need to provide me alot of
info. Which bri package did you install? Perhaps I can install that in a
tmp
Mark Phillips wrote:
Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf
Anyone that tells you to use reinvite= is confused. The option does not
exist (check the source code if you don't believe me). reinvite= is one
of the many Asterisk Urban Myths.
--
Eric Wieling *
Hi all,
We are in the process of selection IP Phones to work with our *new*
Asterisk PBX.
We want to buy 4 for something less than 1000$ but with a nice set of
features to work with our mail box, lines, good sound quality, full
duplex (and maybe speaker phone).
Any suggestions for something
Kevin P. Fleming wrote:
Eric Wieling aka ManxPower wrote:
That's actually not correct. The IP600 supports PoE out of the box.
The IP 30x and 50x support PoE with a special cable from Polycom.
Bummer...I thought the built-in PoE chip was one of the few upgrades in
the 300-301 and 500-501
Hi,
today i have taken a strong look at meetme.c
what i am trying to accomplish is the following:
it should be possible to access an menu from within the conference in
order to perform special tasks, eg. to dial another number so that the
called person is joined with the conderence.
my
On Tuesday 12 Jul 2005 15:51, Colin Anderson wrote:
Thanks for replying. Frustrating, didn't work. Set it to update
automatically, and made an HTML page consisting of:
html
pre
bootloader:
firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin
/pre
/html
Added the HTML page to a
This worked for me:
before compile bristuff edit the file
wcte1xxp.c
near line 1526 initialize the array pci_device_id t1xxp_pci_tbl[]
this way:
static struct pci_device_id t1xxp_pci_tbl[] = {
{ 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board
And if it fails still, check for a buffer overrun on the configuration
file SIPDefault.cnf, the lower firmware versions had less memory
assigned for this file during the upgrade process. Caused me all sorts
of problems :)
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Alexandre Leclerc wrote:
Hi all,
We are in the process of selection IP Phones to work with our *new*
Asterisk PBX.
We want to buy 4 for something less than 1000$ but with a nice set of
features to work with our mail box, lines, good sound quality, full
duplex (and maybe speaker phone).
Any
Half of my 7960G phones work with standard POE, the other half work with the
special rewiring.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kevin P.
Fleming
Sent: Tuesday, July 12, 2005 8:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello list,
does anyone know how to change the interdigit timeout when using Cisco
IP Phone 7940/7960 with SIP-Firmware and Asterisk?
it's default value is 15 sec. but i have nothing found to set this in
tftp-config file etc.
Thanks in advance,
Roland
Hi:
I had an fxo card from Digitnetworks and it was
working fine on my Asterisk box. I then replaced it
with TDM04B. I changed the zaptel and zapata to
include the four channels. When I run ztcfg, I get
configuration errors:
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel
Good day Adam,
I have about 30 Queues configured so at the risk of boring everyone I
have included one of the lines;
exten = _812108,1,Playback(nexus/wel-helpdesk-interwise)
exten = _812108,2,SetCIDName(Client1)
exten = _812108,3,Queue(Client1|Tt|||)
exten = _812108,4,Playback(nexus/im-sorry)
On Tue, 2005-07-12 at 11:04 +1000, Gonzalo Servat wrote:
On 7/12/05, Eric Bullen [EMAIL PROTECTED] wrote:
I hope someone can offer me some help with this. Basically, the current CVS
version of Zaptel will not compile under Fedora Core 4. I have closely
followed the directions in
Interesting part is that I have none G and it worked with 3af
On 7/12/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Michael J. Tubby B.Sc (Hons) G8TIC wrote:
Cisco 7940 and 7960 phones without the G (global) suffix used Cisco
PoE and had their keys labled in local languages. The 7940G and
OK, so I have a nonexistant line in my settings. Why then when I remove
it does my phone call fail?
Rich Adamson wrote:
FYI, there is no such thing as reinvite. Someone started using
that in postings a long time ago and a few people keep posting
it, but it doesn't exist.
(Check
For multi-line without power over ethernet the snom 190 probably fits the bill
and price point. I really like the 360 too, but is probably just over your
price point. We purchased 50 at $225 mark.
Many people are starting to mention the Grandstream 2000 as an option.
For single line the uniden
On 12 Jul 2005, at 15:05, Tony Mountifield wrote:In article [EMAIL PROTECTED],Mark Edwards [EMAIL PROTECTED] wrote: I'd be interested to understand where you read about the 'sequence numbers' issue. This sounds like it might relate to the problem I am experiencing. It's either that, or the DTMF is
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On Tue, 12 Jul 2005, Roland Zagler wrote:
Hello list,
does anyone know how to change the interdigit timeout when using Cisco
IP Phone 7940/7960 with SIP-Firmware and Asterisk?
it's default value is 15 sec. but i have nothing found to set this in
I don't think so.
Your problem seems to do with your not being able to use an
IAX client to transmit DTMF tones properly somehow.
I am using a normal phone to connected to FWD which then
connects to an Asteriskserver using IAX protocol.
The point is that between the phone and the far
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