Frank wrote:
I've been looking through the archives and have not been able to find anyone
with a similar problem but perhaps I'm not searching in the right places. The
problem is that my outbound call sometimes go though and sometimes don't. If
someone can point me in the right direction it
nik martin wrote on 2005/01/19 7:01:
Ronald Wiplinger wrote:
Nabeel Jafferali wrote:
I have provisioned with iaxy.conf:
;
; IAXY Provisioning description
;
dhcp
codec: ulaw
server: 61.220.xx.xx
user: aaabbb
pass: cccddd
register
iax.conf:
=
[623] ; IAXy
type=friend
host=dynamic
Hello,
So far everything that I'm trying with asterisk is working except for this
weird thing. When I try to call voicemail and it asks me for the password I
enter it in but from the debug message I can see that it thinks I didn't
enter anything in. Also when I'm leaving a message it sais press
what endpoints are you using? You probably have a DTMF type mismatch
between asterisk and your endpoint (IP phone or softphone)
-yair
On Wed, 19 Jan 2005 01:49:46 -0700, Tomas Florian [EMAIL PROTECTED] wrote:
Hello,
So far everything that I'm trying with asterisk is working except for this
Hi,
-Original Message-
Did somebody already configured a Digium card on the network
of Versatel in Belgium or the netherlands, and would like to
share his configuration. (zaptel.conf / zapata.conf)
We have HDLC errors (timings i presume)
Yes, we have such setups. Please
Hi,
What do you want to check exacly?
that * is still alive? you want to know the number of concurrent call?
Florian
Hi *,
Does anyone have a lead on a Nagios plugin that speaks IAX or a small
app to do so? I'm trying to set up remote monitoring for my Asterisk
server and only IAX2 traffic is
My solution was to use app_rxfax() and some glue to have faxes
automatically
converted to PDFs and placed in a samba share.
Could you describe more detaily how this could be done. I plan to do
similar thing, so I would like to know which biniries i have to had todo
the same, and to have
an application available called iaxping with would send a set of well
formed iax packets and wait for the response. Unfortunately that
application was written in visual basic, and no source code was
distributed. A skilled coder could probably use some of the required
functions from
On 18 Jan 2005, at 17:23, Michael Devenijn wrote:
Did somebody already configured a Digium card on the network of
Versatel in Belgium or the netherlands, and would like to share his
configuration. (zaptel.conf / zapata.conf)
We have HDLC errors (timings i presume)
Check with your provider to
Hello All,
Want to say that it not latest version of this tool.
Currently I created project on berlios.de called b2bua.
There I will post latest version.
What is this tool? Currently it's AGI script with many asterisk
patches. It support some authorization, accounting (start, stop), LCR,
'Smart'
The login and password are voip/voip
Miguel wrote:
Where I can find the firmware for the Wellgate 3804 ?
The files are:
- 2m4sipfxo.103
- 4fxosip.103
I don't have a password to pick up it at the welltech site.
Kind regards,
Miguel
___
Hi,
-Original Message-
Did somebody already configured a Digium card on the network of
Versatel in Belgium or the netherlands, and would like to share his
configuration. (zaptel.conf / zapata.conf)
We have HDLC errors (timings i presume)
Check with your provider to see
My solution was to use app_rxfax() and some glue to have faxes automatically converted to PDFs and placed in a samba share.
Could you describe more detaily how this could be done. I plan to do
ou can start from here
ftp://ftp.opencall.org/pub/spandsp
I had to manually modify the
I'm setting up a small office PBX on asterisk, and I've got to the
point where I have to decide on
what to do about FAX.
The current situation is: PRI (E1) connected to an E100 and sip hard
phones on their own network.
I'd like to add some very limited fax capability to the system.
Basically
Sending works nearly perfect. It's the receiving that is a pain.
Well, txfax/rxfax is not exactly what we're talking about here.
yes that is right. I have problems sending and receiving
I'm using a v90 modem/fax to test the analog fax from a different
tdm400p port you can hear some click during
The situation is opposite here: I have a Philips fax connceted to an
fxs port of a TDM31B, and receiving from the fax to * with rxfax works
great, while transmitting with txfax *never* works...
did you hear something wrong during the fax handshake? any micro
interruption (click sound) or
Is that excellent work you referred to freely available?
it is released under GPL
I would bet that the community, most especially us, would be most
interested
in it, if you could make it available. Or, better yet, publish it to
something www.SF.net we can all improve upon it in the future. I
The secondary problem I reported earlier (Outgoing MSN andrew not
allowed) seems to have fixed itself. But when I try to call, I get:
*CLI -- Executing Dial(SIP/andrew-4e2d,
Capi/91234567:0412345678) in new stack
-- data = 91234567:0412345678
-- capi request omsn = 91234567
On Jan 19, 2005, at 10:09, Florian Lefeuvre wrote:
Hi,
What do you want to check exacly?
that * is still alive? you want to know the number of concurrent call?
The only think I want to find out is if Asterisk is still alive and
requests coming in via IAX2 are answered. Just some kind of simple
On Wed, 19 Jan 2005, tim panton wrote:
I'm setting up a small office PBX on asterisk, and I've got to the point
where I have to decide on
what to do about FAX.
The current situation is: PRI (E1) connected to an E100 and sip hard phones
on their own network.
I'd like to add some very limited
FYI, that patch was only needed by those that ran * behind a nat/fw box,
and its primary purpose was to reduce registration traffic at broadvoice.
Technically, the patch isn't needed to make * work with bv. (The issue
was that bv would disable an * userid/password if they found your
system to be
it's there already, on http://karlsbakk.net/asterisk/ and under new
plugins on http://sourceforge.net/projects/nagiosplug/
Yes, I've looked at your plugin. However, I'm trying to come up with a
much simpler setup that does not require access to the manager
interface and that does not require
Hi,
We observed the packet size used in asterisk is about
20 ms.
We would like to know if is possible to change this
value to 10 or 30 ms for example.
If so, how could I change it?
Thanks in advance and best regards
__
Do You Yahoo!?
Tired of
Hi,
I'm trying to configure a dual homed asterisk server with iax accepting
connection on all address.
from iax.conf.example
bindaddr = 0.0.0.0 Address to bind to ( all addresses on machine )
but if i register a client using the second ip address i will receive the
response from the first ip
blackburn wrote:
nik martin wrote on 2005/01/19 7:01:
Ronald Wiplinger wrote:
Nabeel Jafferali wrote:
I have provisioned with iaxy.conf:
;
; IAXY Provisioning description
;
dhcp
codec: ulaw
server: 61.220.xx.xx
user: aaabbb
pass: cccddd
register
iax.conf:
=
[623] ; IAXy
type=friend
Hi,
I hope today my 2 fritz cards will be here.
Can you tell me what you did to get the 2 cards working in your system ?
Michiel van Baak
Terrazur
- Originele Bericht -
Van: Andrew Furey
Aan: asterisk-users@lists.digium.com
Datum: Wednesday, 19 January 2005, 10:50
Onderwerp:
I'm trying to do the following:
Several validated users of a web page makes their calls. The call arrive
to Asterisk and is redirected to sip.adiptel.com , where I have only one
user account.
All the callers will arrive to Asterisk with their own user and password
(web validation), and Asterisk
I was browsing the dial cmd page on the wiki, and followed up to the zap
channels page. There are a couple of interesting (ok, so I'm a saddo)
options, specifically the w (wait .5 seconds before dialling the next
digit) and the c option (You may also use the special modifier *c* to
allow for
it's there already, on http://karlsbakk.net/asterisk/ and under new
plugins on http://sourceforge.net/projects/nagiosplug/
Yes, I've looked at your plugin. However, I'm trying to come up with a
much simpler setup that does not require access to the manager
interface and that does not require
Roy Sigurd Karlsbakk wrote:
it's there already, on http://karlsbakk.net/asterisk/ and under new
plugins on http://sourceforge.net/projects/nagiosplug/
Yes, I've looked at your plugin. However, I'm trying to come up with
a much simpler setup that does not require access to the manager
There's always AstWinPeers - which effectively does an IAX2 show peers
or SIP show peers graphically for Winblows.
Oh yeah - I wrote it, so you can find it on the downloads page of the
Daily Asterisk News (in my footer).
The nagios plugin, when probing the manager, counts the number of
Someone have had good luck compilig h323 into YDL?
first thinked was a bug in code but twisted said it is
wierd - isn't that the recursive pthread lib? If so, do you have the
kernel development headers/libs installed?
I've instaled kernel source, what more can i do? any help would be very
I'm going to look into using a network traffic analyzer to capture
such a packet and just use that Windoze iaxping binary to generate
it. I had hoped I would not need to go that far ;)
There:
https://sourceforge.net/tracker/index.php?
func=detailaid=746083group_id=29880atid=541465
Added IAX
Good morning all,
I have a question: if I want to connect an * to a traditionnal PABX via
TDM04B to get PSTN line (incoming and outcoming call), are FXO enough
(no analog phones connected to *)? I just want to add PSTN call (in and
out) possibility for SIP or IAX UA. From what I saw here
Hello,
I am trying to register in asterisk with a softphone (x-lite) and I am
getting the following message:
Jan 19 13:27:01 NOTICE[3359]: chan_sip.c:7531 handle_request: Registration from
'tito sip:[EMAIL PROTECTED]' failed for '192.168.1.5'
In the sip.conf file I have included the following.
Do anyone knows abount European resellers of these products:
* Digium Wildcard TE410P
* CarrierAccess Adit 600
Preferably in Sweden, but Europe is also better.
Have anyone within EU ordered products from these companies directly from the
US?
In that case, how is the service and delivery?
There:
https://sourceforge.net/tracker/index.php?
func=detailaid=746083group_id=29880atid=541465
Added IAX ping :)
Improvement suggestion: The while loop that checks for an IAX answer
currently runs as while (1), so it always runs until the timeout has
been reached. I replaced the while (1)
I have tried uncommenting the section for xlite included in the sample
configuration file sip.conf and I can't register.
[xlite1]
;Turn off silence suppression in X-Lite (Transmit Silence=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
regexten=1234
Daniel Nyström wrote:
Do anyone knows abount European resellers of these products:
* Digium Wildcard TE410P
* CarrierAccess Adit 600
Preferably in Sweden, but Europe is also better.
Have anyone within EU ordered products from these companies directly from the US?
In that case, how is the service
Hello Dominique,
Tuesday, January 18, 2005, 4:28:19 PM, you wrote:
D How are you using the variables in your realtime table?
D Note that you will need to specify variables with the following syntax:
D SIP/${ARG1}|${ARG2} and not SIP/${ARG1},${ARG2} as in extensions.conf
well the problem just
On Wed, 2005-01-19 at 11:25 +0100, Michiel van Baak wrote:
Hi,
I hope today my 2 fritz cards will be here.
Can you tell me what you did to get the 2 cards working in your system ?
http://www.voip-info.org/tiki-index.php?page=Asterisk%20CAPI%20Channels
is, as usual, all you need.
--
Dave
Vahan,
Firmware 103 is working for you?, Not for us.
Pls advise.
Jorge Mendoza
Vahan Yerkanian wrote:
The login and password are voip/voip
Miguel wrote:
Where I can find the firmware for the Wellgate 3804 ?
The files are:
- 2m4sipfxo.103
- 4fxosip.103
I don't have a password to pick up it
That device is complete waste of time and money. I've been contacting
their support for the past 3 months and all I could get were promises
and late replies. Their SIP firmware is not SIP RFC compliant, as it
doesn't follow the Call-ID specification and uses the same one for all
ports.
Jorge
On Tue, 18 Jan 2005 22:11:55 +, nik martin wrote:
I have had very bad experiences with IAXYs so far.. I have pulled them
and will be attempting a refund shortly. Bad audio, overheating and
shutting down until allowed to cool, etc. make it unusable in a business
environment.
That said, is
On Tue, 18 Jan 2005 20:01:51 -0600, [EMAIL PROTECTED] wrote:
Hello all,
We've discovered that VoIP (IAX2) + Citrix + Video is pegging the measly
CPU on the Netopia router our ISP provided. We've got 3Mb/3Mb and will
increase to 4/4 next year.
The Netopia simply breaks out our WAN IPs, and we've
Can anyone help me with this issue? I would greatly any help that can
be provided in resolving this issue.
-Brian
On Mon, 17 Jan 2005 at 14:46 Brian S. Adelson ([EMAIL PROTECTED]) wrote:
I have compiled the latest version of spandsp (spandsp-0.0.2pre10)
against todays CVS head, and I am
Thank You.
At least, we are not alone on this misadventure.
Jorge Mendoza
Vahan Yerkanian wrote:
That device is complete waste of time and money. I've been contacting
their support for the past 3 months and all I could get were promises
and late replies. Their SIP firmware is not SIP RFC
--Original Message Text---
From: Manjit Riat
Date: Tue, 18 Jan 2005 21:15:33 -0800
Clean DocumentEmail false false false MicrosoftInternetExplorer4 /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0;
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Graves
Sent: 19 January 2005 14:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Router Recommendations Please
On Tue, 18 Jan 2005 20:01:51
The problem we're having is transfers don't seem to work? ie: when
someone calls inbound, you drag and drop the call on the extension you'd
like and it just bridges the 2 phones together instead of transfering
the call? Maybe this was intentional or maybe I'm just doing something
wrong? Other than
With Linux you could use CBQ
or HTB script here are the links:
http://sourceforge.net/projects/cbqinit/
http://sourceforge.net/projects/htbinit/
Regards
Ken
-Original
Message-
From: Manjit Riat
[mailto:[EMAIL PROTECTED]
Sent: 19 janvier 2005 00:16
To: 'Asterisk
Just joined the list. Yay.
Seems to me that thanks to VOIP people are now trying to do big things
with still relatively low bandwidth connectivity and/or consumer-grade
hardware. Little Netopia routers aren't going to get you where you need
to be. The solution is to upgrade to either a
I am investigating the use of Asterisk for a new project and am confused
about all the literature available on G.723.1 in pass thru mode.
Specifically, I need to be able to take 2 H.323 channels, each running a
G.723.1 codec, and bridge them together.
However, before I do that, I need to play
Thanks Hanson ... this clear my dudes ;o)
---
Ing. Julio Alvarez Tejera
Unix Trends
*BSD, Solaris Linux
VoIP CT Solutions Finder
Asterisk PBX Consultant
Costa Rica Land +506-359-9753
USA Toll Free +1-888-899-6269
---
extremely stable systems
- Original Message -
Your upstream bandwidth is far too small. Remember, a T-1
(1.5Mbps/symmetrical) is used in a channelized setup to provide 24
64kbps telephone lines (so to speak). Trying to stuff too many calls
into 256kbps using a low bandwidth codec is highly optimistic.
Definitely not something I would do
Linux supports it,
OpenBSD supports it,
FreeBSD supports it
Just pick te OS you like most.
Michiel van Baak
Terrazur
- Originele Bericht -
Van: Michael Graves
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Datum: Wednesday, 19 January 2005, 15:10
Onderwerp: Re:
I've had ordered some items (VoIP starter kit, TDM400P, a couple of ATAs
and an IAXy) from Digitnetworks to Reunion Island (which *technically*
is in the EU despite its remote location) and with UPS shipping it came
fast (about 1 week - but expensive though). No major troubles except
We've
Hi
I have changed my settings to make the whole thing consistent username =
extension = first bit of sip address
found the problem with registration is simply username/password used.
for some reason the setting on phone.cfg under reg.1.auth.userId is NOT
being passed to the phone and is just
On January 19, 2005 12:15 am, Manjit Riat wrote:
My router (1605R) currently does not support QoS. Is there any open source
software available so that I can set one up before the router?
http://www.mixdown.ca/~andrew/dump/rc.tc
I use this on a Linux router with a Sangoma S518 ADSL modem (PCI
I had signed up with BV and wasn't able to call in or out. I had called BV
support and said they wouldn't help if I didn't install the patch. They did
say I was connected to them. Basically I had errors installing the patch so
I asked for a refund. I'm still very new to Asterisk. I did order the
Ok, I think I somewhat understand the concept now that I think about it.
You only want to xfer your trunks correct? In other words Zaptel to
user. We also have IAX trunks for outbound that support multiple lines.
Creating one instance the the IAX trunk won't work because it can only
show one call.
Hello list ,
I´d like to report a success case with a modem based
on chipset : Motorola 62802-51.
It works fine , and zaptel identifies as a X100P
( not clone ) .
Red Alarms can be identified . :) This doesn´t
occurred on MD3200 ambient chipsets.
Best Regards ,
--
- Jefferson Carvalho
Analista de
Problem solved :
The reason was quite simple ... but annoying :
Interrupts !!! damned !!!
Thank you
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens Florian Overkamp
Verzonden: wo 19/01/2005 10:08
Aan: 'Asterisk Users Mailing List -
Versatel uses CRC4 (in Belgium)
zaptel.conf :
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
alaw=1-31
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens Florian Overkamp
Verzonden: wo 19/01/2005 10:33
Aan: 'Asterisk Users Mailing
Figured out that with the 7940's (unlike the 7960's for somereason)
you have to specify the proxy_register: 1 option in the config file
for the phone, or in SIPDefault.cnf. This fixed the problem for us.
-Chris
On Wed, 19 Jan 2005 13:05:00 +1300, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
On 19/01/2005 22:08 Jorge Mendoza said the following:
Thank You.
At least, we are not alone on this misadventure.
Jorge Mendoza
Vahan Yerkanian wrote:
That device is complete waste of time and money. I've been contacting
their support for the past 3 months and all I could get were promises
and
I've got a simple MeetMe conference configured using Asterisk 1.0.3 on
Gentoo. I'm using zaprtc for timing from the bri-stuff package.
extensions.conf
exten = 37455,1,NoOp(Drill Squad Conference)
exten = 37455,2,Monitor(wav,drillsquad-37455,mb)
exten = 37455,3,MeetMe(37455,pMs)
Now, when I
Hello
I have a PBX for our company with in front a ISDN * ISDN server.
I would like to detect if the outgoing line is a fax call or voice call.
I know ther's a fax detection in zaptel.
Now the question can I make desision in extentions.conf when * detects a
fax call.
FAX PBX ZAP/G1 ZAP/G2
Their server seems to be down though...
- Original Message -
From: Wilson Pickett [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Wednesday, January 19, 2005 3:26 PM
Subject: Re: [Asterisk-Users]
Now, I would very much like to remove the canreinvite=no from the
provider's definition on sip.conf, but doing so causes Asterisk to
send
a re-invite to the provider pointing to a private IP. I thought that
correct localnet entries would solve this...
By changing to canreinvite=yes,
hi
I really want a good queue_log analyser, but I don't want to waste
EUR 1000 for something like that, so I thought starting a small project
for it. I started off in php just creating a basic parser for the log,
and I'll go on extending it. See
On 2005.01.19 01:39 tim panton wrote:
My options include;
1) get a basic fax machine and plug it into a (iaxy/sipura?)
ATA. Configure the ATA to do alaw
so asterisk doesn't have to do any transcoding and hope it works.
2) use spandsp on my asterisk system and add suitable print
queues
3) get
There's a few (open source/free) ones in development. I myself am
developing one of them.
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd
Karlsbakk
Sent: 19 January 2005 15:33
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Thats amazing! Worked like a charm...any explanations as to why this happens?
On Wednesday 19 January 2005 03:21 am, Matt Riddell wrote:
Frank wrote:
I've been looking through the archives and have not been able to find
anyone with a similar problem but perhaps I'm not searching in the
Is it possible to have asterisk play some short audio files to a running
conversation?
Thanks
Jens
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup
On Wed, 19 Jan 2005 15:44:44 +1100, Adam Goryachev
[EMAIL PROTECTED] wrote:
On Tue, 2005-01-18 at 20:03 -0600, Eric Rees wrote:
Has anyone been able to find a way to disable call-waiting on Polycom
phones?
I've not yet found any
Title: Message
Hi,
Asterisk ${EXTEN} variable hold only first 79 characters of the
extension.
Is it
possible to do some modifications in order to get entire
extension?
Regards,
Stojan
Sljivic
___
Asterisk-Users mailing list
'morning everybody,
Here is the setup: 5126800422 called 3035 (3035 is a Cisco 7960). The call
is g729. 3035 presses 'Conference' on her phone and calls 8327549222. This
call is ulaw. (65.72.107.2 is our Cisco 7206 SIP-PRI gateway.)
asterisk*CLI sip show channels
Peer User/ANR
Hi All,
For a small installation using ITSPs via DSL is G.729 a worthwhile
exercise? I have G.729 capable SIP phones and my ITSPs cupport the
codec so I could go end-to-end without transcoding. What's call quality
like compared to G.711, GSM or iLBC?
Michael
--
Michael Graves
Lee Howard wrote:
On 2005.01.19 01:39 tim panton wrote:
My options include;
1) get a basic fax machine and plug it into a (iaxy/sipura?) ATA.
Configure the ATA to do alaw
so asterisk doesn't have to do any transcoding and hope it works.
2) use spandsp on my asterisk system and add
Dinesh,
Thank You for your comments.
The problem is not only ports registration, but many others issues.
For instance, with last version 103, the hunting feature does not work
(a least for us). If we enable all 4 ports, with an outbound call, the
first call seize the port 2, the second call got
[EMAIL PROTECTED] wrote:
Hi All,
For a small installation using ITSPs via DSL is G.729 a
worthwhile exercise? I have G.729 capable SIP phones and my
ITSPs cupport the codec so I could go end-to-end without
transcoding. What's call quality like compared to G.711, GSM or iLBC?
Low bandwidth
On 2005.01.19 01:39 tim panton wrote:
My options include;
1) get a basic fax machine and plug it into a (iaxy/sipura?)
ATA. Configure the ATA to do alaw
so asterisk doesn't have to do any transcoding and hope it works.
2) use spandsp on my asterisk system and add suitable print
Has anyone been able to find a way to disable call-waiting on Polycom
phones?
I've not yet found any solution to this, and I haven't seen anyone else
who has. Definitely please let us all know if you do find the answer...
I wrote to Polycom about this a couple of weeks back, but I haven't
heard
I want to know if there is way to break out of the voicemail message.
for example:
On my Noterl PBX when you dial you number from any where
you get your recorded voice mail message, but during the message I
press 81 and break out of that message. It then
prompts me for my PIN thus allowing me
about buggy firmware I want to know what I'm getting into before upping past
my current 1.0.5.11.It's relatively stable, and the last thing I want to
do is update to a flaky firmware
I don't know why, and Grandstream *swears* I'm the only one having
this problem, but I am stuck with
If you put the following in your Dialplan, pressing * should break you
out of voicemail and call VoiceMailMain
exten = a,1,VoicemailMain,EXTEN
exten = a,2,Hangup
On Wed, 19 Jan 2005 11:33:23 -0500, kurt x [EMAIL PROTECTED] wrote:
I want to know if there is way to break out of the voicemail
Have you checked the firewall on the server/router ? Is there a way to
ping from a Grandstream to the server ? Just a thought...
On Wed, 2005-01-19 at 17:38 +0100, Wilson Pickett wrote:
about buggy firmware I want to know what I'm getting into before upping past
my current 1.0.5.11.
18 - 22 Kbps my dream!
I have asterisk - INTERNET - asterisk connection with IAX2 and I try
iLBC, gsm, g729 and speex and the minimun bandwidth was 38 Kbps for 1
channel.
What the parameters do you set to have this rate ???
Thank you.
Miguel Ruiz Velasco Sobrino wrote:
I have an
I'm still waiting for them to release a firmware update that shows caller id
name and number at the same time while the phone is ringing. That should be
submitted as a feature request as well I assume.
- Original Message -
From: Noah Miller [EMAIL PROTECTED]
To:
any suggestion of a internal fax-modem type that works
well with Asterisk?
__
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com
___
Asterisk-Users mailing list
On Jan 19, 2005, at 10:03 AM, Peer Oliver Schmidt wrote:
Lee Howard wrote:
On 2005.01.19 01:39 tim panton wrote:
My options include;
1) get a basic fax machine and plug it into a (iaxy/sipura?)
ATA. Configure the ATA to do alaw
so asterisk doesn't have to do any transcoding and hope it works.
-Original Message-
From: Roy Sigurd Karlsbakk [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 19, 2005 6:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk monitoring with
Nagios and IAX (RoySigurd Karlsbakk)
Rich Adamson wrote:
Let me restate my problem. I have a group of users behind a constrained
pipe to the public network. There are a few mobile users that will
mostly be working from their home offices. I *really* want to avoid
having a call from a mobile user to a public number cause double the
I have found that having multiple entries in the DNS sometimes seems to
cause a problem with 1.0.6.x versions. Try using one DNS entry, preferrably
a nearby public one (e.g. your ISP's).
Michael Crown
Managing Partner
The VoIP Connection
--
Message: 15
Date: Wed, 19
Low bandwidth
Low CPU utilization
Best audio quality
I think you might want to clarify that Best audio quality is in relation to
other highly compressed codecs. Certainly my (albeit limited) experience is
that g711 is much more clear than g729. Compared against gsm, for example,
however, the
Hi List!
I have an interesting problem. I am behind a NAT Firewall which works
fine with SIP. I am connected to T-DSL in Germany and there the
DSL-Connection is interrupted every 24hours and buck a few seconds later
with a new dynamic IP.
My Asterisk is registered with several SIP-Providers and
Hi,
I'm in big trouble I've got two serwers with Asterisk 1.03 both serwer with 2
ISDN cards. When I call isdn server card and then choose other msn (disa)
I can make connection with other phone (I've got standard ISDN PBX).
But echos are horrible.
My configuration:
Athlon 2.0
256 RAM
2 HFC
Dear Claudio,
I'm testing the welltech gateways (3804 firmware 4fxosip.102) and I am trying to make the Asterisk answer the calls from 3508 directly (with 2nddial off) it means throw hotline service.
Do you know how to make the Asterisk answer a call from :pstn-to-3508 and 3508-hotline-Asterisk
Could you post your sip.conf and /tfpboot/* files? (wiht passwords removed)
off list would be fine. I'll take a look at them and see if I can find
anything.
I just got my 7960G working. I love the phone, but they were clearly
designed to be managed in large groups and setting up one is nearly as
1 - 100 of 209 matches
Mail list logo
