I have downloaded iaxmodem and gone through the readme but not yet installed
it. I currently use rxfax to receive in the vicinity of 1200 faxes per day
and 5000 or more pages (faxes vary from single page to 30 pages) per E1,
with a peak load of about 12 concurrent inbound faxes to rxfax. Best
I've been having a lot of problems with Broadvoice lately. Anyone else
been without service for extended periods of time this week?
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
--- Nate Kapi [EMAIL PROTECTED] wrote:
I've been having a lot of problems with Broadvoice
lately. Anyone else
been without service for extended periods of time
this week?
Service is down right now
__
Yahoo! Mail - PC Magazine
Hi,
for the snom360 it is working the same way. Use firmware version 4.3 and be
aware that the message is send to a specific SIP line and the phone is
displaying it only if this SIP line is the current active line (outgoing
identity) symbolised by a black phone (snom360) or the text in
Yes it will support it, you should look up HDLC on the wiki...I went
through this a year ago and had a hard time setting it up. It might be
easier now though. I would recommend going another route and getting
the data brought in seperately with it's own router. You'll also have
better redundancy
I have 4 * servers interconnected with IAX trunks. Three are on a local LAN,
one is accessible over a VPN tunnel out of the office. The IAX peer status
(iax2 show peers from the CLI) will sometimes show upwards of 300ms.
Considering the lag and distance, I am not entirely surprised.
Anyway - my
I guess the 2U is not bad... Im going to call supermicro and check what they
have. What kind of CPU are you using guys? Seems supermicro has everythiung
except the CPU and the HD right?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Kevin Bockman
trixter http://www.0xdecafbad.com wrote:
On Wed, 2005-10-12 at 17:46 -0700, Paul Mahler wrote:
You need about 30MHz per channel. That means the Soekris can only handle part
of a T1, it will never handle a quad span.
Paul
How was that determined?
I have a problem
On Thu, 2005-10-13 at 14:03 +0800, Craig Guy wrote:
I have downloaded iaxmodem and gone through the readme but not yet installed
it. I currently use rxfax to receive in the vicinity of 1200 faxes per day
and 5000 or more pages (faxes vary from single page to 30 pages) per E1,
with a peak
VPN?
IAX and an SSH Tunnel?
- Original Message -
From:
Kellner,
Peter
To: asterisk-users@lists.digium.com
Sent: Thursday, October 13, 2005 5:18
AM
Subject: [Asterisk-Users] Wanting to Make
a PocketPC have a secureConnection to asterisk server
Does
What excatly does it do?
What messages does it send out?
And what software needs to be configured to listen for these messages?
Answer these questions and maybe more people will download the source :-)
Steve
(Not being an arse just reckon a better description is needed)
- Original Message
Hello Pedro,
you should do this using agent priority groups; this way first all low
priority agents are filled, then another group is used up.
Thanks
l.
On Wed, 12 Oct 2005 19:30:43 +0200, Pedro Nunes [EMAIL PROTECTED]
wrote:
Hi there,
Does anyone know how to setup an overflow queue?
Armin Schindler wrote:
On Sat, 8 Oct 2005, Cedric Fontaine wrote:
[logfiles]
console = notice,warning,error
I don't mean the capi.conf. Do you have an extension in context 'entrant'
that matches 9100 ?
So I added it in the logger.conf and you were right... There was a
problem with matching
Hi all,
Does anybody has good
working solution for email to fax (simply sending faxes) by asterisk.
Thanks,
Bob.
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Asterisk-Users mailing list
Asterisk wasn't correctly identifying that the file is
actually wave49. We logged into your server and fixed it.
-Mike
Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728
ext. 611 sip:[EMAIL PROTECTED]
From: Tom Vile [mailto:[EMAIL PROTECTED]
Sent: Tuesday,
On Thu, 2005-10-13 at 08:16 +0100, Steve Daniels wrote:
VPN?
IAX and an SSH Tunnel?
Does anyone know of a good solution to create a secure
(encrypted) connection from a pocketpc (IPAQ 6515 in my case)
to an asterisk server?
Pocket pc supports VPNs natively.
I have lost the password of the telephone, so I must
do a reset of the telephone. How can I do?
I have a VOISMART telehone: IP PHONE 106
Thanks
___
Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB
What version are you using?
Try SetCIDName(Fred)
Check voip-info's wiki
HTH
Steve
- Original Message -
From: Serge Lhermitte [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 12, 2005 5:57 PM
Subject: [Asterisk-Users] AGI and set_callerid for number and
Leaving apart people like Mr Totaro, always speeking about other men
without knowing them ( I am a consultant, I am in the Networking since
early '90, I never ask anything about NAT..) anyway I am going to put
that book in my basket (i have a subscription with Oreilly, I can access a
certain
On Thu, 13 Oct 2005 00:39:06 +0200, Tom Rymes [EMAIL PROTECTED]
wrote:
What we have done is to set up a single queue that all calls come into.
For the agents that we want to be our Front Line (i.e.: Customer
Service Reps), we give them a penalty of 0. Our Overflow group (i.e.:
Customer
On Thu, 2005-10-13 at 08:20 +0100, Steve Daniels wrote:
What excatly does it do?
What messages does it send out?
And what software needs to be configured to listen for these messages?
Answer these questions and maybe more people will download the source :-)
As was explained to me via
Using SIP? IAX?
One way sound is usually a SIP and nat/firewall problem, make sure ports are
forwarded.
Steve
- Original Message -
From: Peter Ankerstål [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 12, 2005 10:39 PM
Subject: [Asterisk-Users] Maximum
Seth Remington wrote:
Hello All,
I saw an add in my latest Linux Journal advertising Sangoma's new AA
series of FXO/FXS analog cards with on-board echo cancellation, but I
can't find any information at all on them. Even the link given in the
advertisement is a dead end as far as I can tell.
What kind of connection? Voice or
Configuration?
SIP? IAX? ssh? I have an
ssh client for PPC in one of my archives somewhere. Sorry, dont
know what its called, where its from, its been a while since Ive needed
it. But it does exist.
If thats what you need, Ill
take a look for it on
Hi Bob,
I've justed looked at inter7 solution and perhaps that is what you're
looking for (http://www.inter7.com/?page=astfax)
Greetings Otto
Hi all,
Does anybody has good working solution for email to fax (simply sending
faxes) by asterisk.
Thanks,
Bob.
Thanks a lot.
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedro
Nunes
Sent: mercredi 12 octobre 2005 19:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] AGI and set_callerid for number and name
Curse,
On Thu, 2005-10-13 at 09:18 +0200, Bohuslav Coufal wrote:
Hi all,
Does anybody has good working solution for email to fax (simply
sending faxes) by asterisk.
Effectively T.37 does that, however what you prolly wanna look at is
hylafax to process the emails (perhaps by procmail). From
I'm trying to figure out what an appropriate deployment model might be.
Whether to have iaxmodem installed on the hylafax server with a switched
ethernet connection for iax2 to the * server with the PRI, or to have the
iaxmodem on the PRI * server and channel the tty comms across the network.
Thanks, I'll try it.
Bob.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, October 13, 2005 9:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Email to FAX
Hi Bob,
I've
On 10/12/05, Mir [EMAIL PROTECTED] wrote:
We have discovered a problem with DTMF on Asterisk.We have a setup with a T1 from PSTN going into an Asterisk box, and
then out again on T1 and into a normal PBX (EADS)We use it to record all calls going to/from the PBX.The problem is that when we
Thanks,
That will fix my problem... And agent skills, is that possible too??
Thanks again
Pedro Nunes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lorenzo
Emilitri
Sent: quinta-feira, 13 de Outubro de 2005 8:17
To: Asterisk Users Mailing List -
Thanks,
That will fix my problem... And agent skills, is that possible too??
Thanks again
Pedro Nunes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes
Sent: quarta-feira, 12 de Outubro de 2005 23:39
To: Asterisk Users Mailing List -
On Thu, 2005-10-13 at 15:55 +0800, Craig Guy wrote:
I'm trying to figure out what an appropriate deployment model might be.
Whether to have iaxmodem installed on the hylafax server with a switched
ethernet connection for iax2 to the * server with the PRI, or to have the
iaxmodem on the PRI
Just remember to set your phone in the group with the highest possible
priority :)
On Thu, 2005-10-13 at 09:36 +0100, Pedro Nunes wrote:
Thanks,
That will fix my problem... And agent skills, is that possible too??
Thanks again
Pedro Nunes
--
Trixter http://www.0xdecafbad.com
Hi,
when I try to send fax by example in README I got nothing. On asterisk console
i saw this:
-- Attempting call on Zap/4/585228796 for application
txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1)
Channel Zap/4-1 was answered.
Launching
Hi list,
I have a Wildcard TDM400P REV I Board 1 with 4 FXO
modules and * 1.0.9 up-and-running.
Only 2 FXO ports are used for 2 analog phones and
are doing fine.
I now wanted to use the 3rd and 4th port, but when I
insert an analog phone, take it off hook, I do not
get a dial tone.
With my
On Thu, 2005-10-13 at 10:45 +0200, Coufal Bohuslav wrote:
Hi,
when I try to send fax by example in README I got nothing. On asterisk
console
i saw this:
-- Attempting call on Zap/4/585228796 for application
txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1)
Hello all,
I try to use a agi script to get a variable from * und put them into a
script which gives me another variablke and put this in *.
My problem is now it seems the var ID is empty coz i always jump into
the result 0 loop.
The $MSN should be in the SetCIDNum.
#!/usr/bin/php -q
?php
Yes, using sip. The ports are forwarded. The calls going to the other asterisk
server works fine. The problem occurs only when people who are registred to my
server tries to call.
On Thu, 13 Oct 2005 08:30:17 +0100
Steve Daniels [EMAIL PROTECTED] wrote:
Using SIP? IAX?
One way sound is
But it seems that Asterisk understand that he has to dial (the dialed number
is correct),
-- Attempting call on Zap/4/585228796 for application
txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1)
it seems that zap channel had answered (but nothing to try dial),
Actually we 're running the sip protocol but in the past we did also use
h323 in combination with tedas phoneware server (german voip solution). Both
ran on SmartNode side very stable. Caller ID Name with sip/h323 should not
be a problem, but here in Germany I'm not really shure, if the telco
Yeah I missed that in the original, sorry bout that.
are you sure that the other end didnt hang up? You may want to test
this by calling a number you have access to so that you can at least
rule that out.
The only other thing I can think of is that txfax itself is aborting and
returning
Hello
Im getting this error any body have any idea how to fix it
pbx_spool.c:229 attempt_thread: Call failed to go through, reason 3
Regards
Fahd Ansari
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Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog
line, and FXS port is for connect analog phone. Are you sure that in 3rd and
4th ports you have immediate=no?
regards,
srsergio
-Mensaje original-
De: Alex Ongena [mailto:[EMAIL PROTECTED]
Enviado el: jueves,
On Thu, 2005-10-13 at 12:44 +0200, Sergio Serrano wrote:
Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog
line, and FXS port is for connect analog phone.
sorry, my mistake, these are all FXS ports with FXO signaling. I always
mix them up.
Are you sure that in 3rd
Hi,
Can anyone recommend a USB phone that can be used under Linux, either
interfacing directly with Asterisk in some way, or using a soft phone
program on Linux that doesn't need screen interaction (only using the
phone's keypad)?
The idea is to be able to plug it into the USB port of an
On Thu, 2005-10-13 at 12:44 +0200, Sergio Serrano wrote:
Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog
line, and FXS port is for connect analog phone. Are you sure that in 3rd and
4th ports you have immediate=no?
if it may help,
I could just stop *,
# rmmod wcfxs
Yes, i am having timeouts on registering to the LAX sip server of
broadvoice.
Marco.
Nate Kapi wrote:
I've been having a lot of problems with Broadvoice lately. Anyone else
been without service for extended periods of time this week?
___
Check your Revision card, if it is Rev H in zaptel sources you have a
zconfig.h with a Flag to Revision H. Try it.
regards,
-Mensaje original-
De: Alex Ongena [mailto:[EMAIL PROTECTED]
Enviado el: jueves, 13 de octubre de 2005 12:56
Para: Asterisk Users Mailing List - Non-Commercial
My asterisk is purely connected to the outside world via SIP.
When I use Dial() with the m-option, that should ensure music-on-hold,
it works perfectly as long as I am calling a SIP number, but when I
call a mobile phone, the music-on-hold disappears.
Any ideas on the cause of this or how to fix
Hi!
Has anyone tested this IAX ATA?
Their free softphone is GREAT
https://www.virbiage.com/products.php
Regards
Anders Svensson
___
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Asterisk-Users mailing list
My number is not submitted.
I updated my asterisk but this error still occurs coz of the in the
SetCallerID tag thats why it will be a empty SetCallerID is submitted.
Is there a fix to correct this error?
-- Executing SetCIDNum(SIP/31-752a, 4989427) in new stack
-- Executing
Has anyone tested this IAX ATA?
https://www.virbiage.com/products.php
For some reason, their IAX hardphone was coming soon for two years on
the site and then... still no word.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
Hi
Sangoma a104 card have in product specyfication support for
Line protocol SS7 ,
http://www.sangoma.com/products/p_aft-104-specs.htm
[..]
Line protocols
Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC.
[..]
Anyone of you guys use line protocol SS7 for E1/T1 termination in
Hi,
Is there a command to start simpleswitch from an extension? For
example it would allow me to dial in to my * box and get a dial tone to
make an outgoing call.
Thanks,
Derek
--
Derek Conniffe
Rivertower Ltd
DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Freephone)
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DISA is probably
what you need.
Thanks,
Steve Totaro
- Original Message -
From: Derek Conniffe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October
René Enskat [Teamware GmbH] wrote:
My number is not submitted.
I updated my asterisk but this error still occurs coz of the in the
SetCallerID tag thats why it will be a empty SetCallerID is submitted.
Is there a fix to correct this error?
-- Executing SetCIDNum(SIP/31-752a, 4989427)
On Wed, 12 Oct 2005, Jason Walker wrote:
I have 4 * servers interconnected with IAX trunks. Three are on a local LAN,
one is accessible over a VPN tunnel out of the office. The IAX peer status
(iax2 show peers from the CLI) will sometimes show upwards of 300ms.
Considering the lag and
Hello,
I have a polycom ip600 and eyebeam. When I call from polycom to
eyeBeam, everything, including audio works. When I call the other side
(from eyeBeam to polycom), I get no audio. In both cases, eyeBeam shows
the same codec: g711u. Also sip show channels shows ulaw codec for both
sides and
Tony Mountifield wrote:
Hi,
Can anyone recommend a USB phone that can be used under Linux, either
interfacing directly with Asterisk in some way, or using a soft phone
program on Linux that doesn't need screen interaction (only using the
phone's keypad)?
The idea is to be able to plug it into
Try disabling inband call progress tones. Let Asterisk handle everything.
In sip.conf add the line:
progressinband=no
On 10/13/05, Lars Dybdahl [EMAIL PROTECTED] wrote:
My asterisk is purely connected to the outside world via SIP.
When I use Dial() with the m-option, that should ensure
Hello everyone,
I have been asked for "directed pickup" and saw
that both "PickupChan" from bristuff and "Intercept" applications
do the dirty work.
I have tried both on asterisk-1.0.9 (
BRIstuffed-0.2.0-RC8o ) but I always got an error when trying to pick the
ringing call.
the debug says:
Are you looking on purchasing one?
francis
www.VoIPware.ca
On 10/13/05, Anders Svensson [EMAIL PROTECTED] wrote:
Hi!
Has anyone tested this IAX ATA?
Their free softphone is GREAT
https://www.virbiage.com/products.php
Regards
Anders Svensson
hi,
do you someone know tool that can get data like
latency/bandwith/jitter/packet loss (in one program)
- it must be functional behind nat
- multiplatform (AJAX,java applet)
- preferably on SIP and IAX ports
- can be client/server
- easy to use ;)
---
Title: Patrick Briefpapier
Hi
Martin,
I saw your problem
listing on the Asterisk mail archives. I seem to have the same problem with the
ISDN 'lacking dialtone' message
I still have not
been able to get it working, could you share your modem / extension / sip conf
files?
Thanks in
Craig Guy wrote:
I have downloaded iaxmodem and gone through the readme but not yet
installed it. I currently use rxfax to receive in the vicinity of
1200 faxes per day and 5000 or more pages (faxes vary from single page
to 30 pages) per E1, with a peak load of about 12 concurrent inbound
Hi all!
I've got a problem with thia PA168S/AT320P telephone.
I got 2 servers: one with SER and the other with Asterisk.
All users are on SER and Asterisk is the gateway/voicemail.
In these days I'm starting some tests using Asterisk accounts users.
With this PA168S/AT320P, if I use it with a user
John Lange [EMAIL PROTECTED] wrote:
My apologies for the cross-posting.
If you think you should apologize for it, don't do it. If you think
it is okay to do it, don't apologize.
Doug
--
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
Also take a look at www.trustfax.com
They've done a fine job for us and have several different plans that
address from very low to high volume faxing. Receiving faxes via email
as pdf files is great, very timely, with no errors identified in the
past six months.
From:
1)What is the protocol you are using? SIP or IAX2?
2)Have you applied the correct firmware to the Phone?
Pa168 phones are falwless when connecting to Asterisk.
Start the configuration as asimple entry as under.
I have added Port address and allowed codecs in the config below:
[221]
hy all i want to knwo that which voip fone( hard fone
) will be better either it should be iax, sip or h.323
( that should be good and not too expensive ) i
want to have a setup of 200 fones in five offices.
and is there any card available to connect four pstn
lines. like in single channel
Hi,
Iperf does it, but is not made for running as MRTG or Nagios.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of marek cervenka
Sent: Thursday, October 13, 2005 9:12 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] link quality monitor
Hi, please add me to the mailing list
I also can donate webspace, bandwidth, IAX local dialtone to 780 area code,
and DNS services.
btw how are you going to do the conference call, with MeetMe?
-Original Message-
From: John Lange [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 12,
Hi,
Is there a script anywhere which would import existing *.conf entries
into a mysql database for use with the realtime architecture?
Thanks in advance.
--
-Barry Flanagan
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Ok so I've just built and installed a CVS (HEAD) version of asterisk
on RHFC2 running a 2.6.13.3 kernel.org kernel. I installed the samples
via make samples. Everything seems to work except one thing. I'm
trying to do the connect to the Digium IAX demo server portion of the
demo (dial 500) and
Yes I was interested to
test them. They are not available on the link you submitted either
Anders
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francis Ballares (VoIPware.ca)
Sent: den 13 oktober 2005 15:12
To: Asterisk
Users Mailing List - Non-Commercial
Hi there:
I have a simple question...can I use the internal mail server that
uses * as my organization pop-smtp server, if so how can I do it. Thanks
Hector
___
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Asterisk-Users
Hello
I need Moscow dids urgently,
Contact me offline [EMAIL PROTECTED]
Regards
Mehdi Chouikh
Universal Telecom
Spain
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Im wanting both the voice and the
configuration to be secure. (very secure). I dont care if it is SIP or
IAX but I do need a softphone on the pocketpc I can use. Id appreciate
if you could take a look this weekend for me.
Thanks, -Peter
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
for some reason your script is not executing the get_var correctly, as
you can see in the output, asterisk is saying: invalid or unknown
command.
check the internals of your script, the most common reason is that you are mispelling the command.
best regardsOn 10/13/05, René Enskat [Teamware
hi,
is there anyway to make * to detect callerid before first ring.
i know that it seems silly; but here i have a case that Telco sends
the caller-id before first ring. this issue is detected by installing
a callerid detection device on the line. it shows callerid just before
the first ring. so *
I did try this and did get it to register as this peer.
However inbound calls to that number are still coming into
the context defined in [general] sip.conf
I now have two numbers configured, the new peer as you sugested
and my original that just has the register line
without an associated peer
Hi,
thanks to reply:
1)SIP
2)yes. I've used the original 1.46 for SIP protocol
Also your solution do not work.
Are 2 days that I'm trying configurations and googling for this
problem, but nothing!
Always: LOG ON FAILED
I've saw about problems with this phone, but my hope was that with the
new
Dear sirs,
I believe that this question should go to Steve Underwood,
but if someone else also has something to say, I have my ears totally open.
After differents tests (None of them worked), Im
ready to install spandsp, app_txfax app_rx fax to try fax to email
email to fax.
I just tested it and it's working fine.
Does your Linux box have internet access?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
It's a REV I ...
Txs
On Thu, 2005-10-13 at 13:06 +0200, Sergio Serrano wrote:
Check your Revision card, if it is Rev H in zaptel sources you have a
zconfig.h with a Flag to Revision H. Try it.
--
NEW: aXs GUARD hands-on Trainings v.7.0
more info at http://www.axsguard.com/indextraining.htm
Matt Riddell wrote:
I just tested it and it's working fine.
Does your Linux box have internet access?
Yep, but through a firewall. I figured it probably works ok and
that I must just be doing something wrong. The only config file
I changed was sip.conf. In this file I just uncommented out
I have other IAX ATA's available at VoIPware.ca - I have tested them personally and they work great.
thanks,
Francis
www.VoIPware.ca
On 10/13/05, Anders Svensson [EMAIL PROTECTED] wrote:
Yes I was interested to test them. They are not available on the link you submitted either
Anders
I want to modify the info
Libtiff is 3.5.7 (uninstalled the 3.7.4 and install this one
after reading a note about the crash)
Audiofile is 0.2.6
Thanks,
Carlos Alperin
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DISA(password|context)
On Thu, 2005-10-13 at 12:58 +0100, Derek Conniffe wrote:
Hi,
Is there a command to start simpleswitch from an extension? For
example it would allow me to dial in to my * box and get a dial tone to
make an outgoing call.
Thanks,
Derek
have you configured the STUN server on the phone to any one of the
available stun servers like stun.xten.net?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of FaberK
Sent: Thursday, October 13, 2005 10:40 AM
To: Asterisk Users Mailing List -
Anyone have any
ideas as to why a call coming in won't ring the phone? I can call the
phone from my cell and when I hear it ringing on the cell phone I pick up the
house phone that should be ringing and am able to talk. I have tried two
different pap2-na adapters, have verified the ports on
Right now, but nothing changed.
2005/10/13, Kanuri, Seshu (Company IT) [EMAIL PROTECTED]:
have you configured the STUN server on the phone to any one of the
available stun servers like stun.xten.net?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I have no clear idea how many people actually use my software for fairly
high volumes. There are now clearly many thousands successfully using it
for modest levels of faxing. I have heard from a few people doing rather
higher volumes than you. Other people have problem - I mean genuine
Craig Guy wrote:
I'm trying to figure out what an appropriate deployment model might
be. Whether to have iaxmodem installed on the hylafax server with a
switched ethernet connection for iax2 to the * server with the PRI, or
to have the iaxmodem on the PRI * server and channel the tty comms
Why don't u attach the setup page of the phone ?
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di FaberK
Inviato: giovedì 13 ottobre 2005 17.56
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [Asterisk-Users]
I have 2 * boxes.
1 has 2 PRI's from the Telco, and a PRI to the 2nd *
The other has ZAP channels to Channelbanks for endusers.
If someone on the second box calls a Toll Free number (it probably
doesn't matter that it is toll free) that is auto answered by an auto
attendant (QVC, a Bank, the
Hi,
I have an asterisk box with a TE410P (quad pri) which has 3 spans in
use, 1 and 3 to two different telcos, span 2 to a legacy Norstar MICS.
Everything has been working fine for months, but early this morning,
the 1st span stopped accepting incoming calls, but outgoing calls on
this span still
I have been seeing the subject behavior on head for a few days now..
(been trying nightly builds to see if a bug causing this has been fixed)
on a sip show channels I get a little of active channels that I can
correlate calls to.. but I also have some dead channels listed that
should no longer
Sounds similar to a problem I've seen with a slightly different
setup Calls to certain AA/PBXs were not passing
progress information beyond 10 seconds into the call. Can
you check your logs for the exact amount of time after the setup that
the call gets dropped? I'm guessing you'll see 10 or
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