Hello
On Fri, 14 Oct 2005 01:25:20 -0500, Kevin P. Fleming wrote
> Marco Balmer wrote:
> > Any ideas or hints?
> Yes. Whatever documentation told you that you could share a Realtime
> SIP peer database between two Asterisk servers was in error (or at
> least very incomplete).
Server1 acts as a
I think, that mistake is between PC and chairs. When i have not outgoing
lines it's too hard to call out. Now i'm in state, that example form
README dialed and i'm trying to receive fax on other side.
Thanks,
Bob.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Be
I have very loud sound through IAX2 channel, very saturated in some
moments.How to find where is problem. I think problem is at provider
side, but how to be doubtless?
Is there any method to measure and change sound level on IAX channel (like
on Zap channel)?
__
Marco Balmer wrote:
Any ideas or hints?
Yes. Whatever documentation told you that you could share a Realtime SIP
peer database between two Asterisk servers was in error (or at least
very incomplete).
There are ways to do it right now, but it's not trivial and does not
provide all the func
Peder @ NetworkOblivion wrote:
And it's wink-start on an E&M analog circuit, not on a standard analog
phone line from your telco. You would need a card that supports E&M to
do it even if the telco provided it (not sure if the Digium cards
support it, but I tend to doubt it).
We do not have a
Hello,
how i can tranfer call to another user? Im using X-Lite, i have configured in
features.conf:
[featuremap]
blindxfer => #1
disconnect => *0
automon => *1
atxfer => *2
But when im dial *2 in conversation nothig happens.
What can br problem?
Im using asterisk CVS-HEAD from 02/09/05.
Re
Hello @ all,
I hope you can help me.
server1: asterisk-cvs HEAD 2005-10-13
server2: asterisk-cvs HEAD 2005-10-13
I've configured RealTime (sipusers) on server2 together with a MySQL database.
The account in the database exists. It seems to be configured right. Then I
can read realtime infos with
I'm running into errors when using Enum lately. I can't figure out what
the problem might be as I've had Enum up and running in the past. I'm
running the latest CVS-Head compiled version. I've also tried using the
new Enum function with the same results. When doing a lookup on a number
that
All,
I have a TDm04b card an 8 SIP extensions. Calls come into the TDM and
are answerd by the auto attendant. When an extensions is entered I see
the Dial(SIP/100) on the console but the phone never rings...
I can pick up any extension and call 100 and it rings just fin>e and they
answer and
All,
I have a TDm04b card an 8 SIP extensions. Calls come into the TDM and
are answerd by the auto attendant. When an extensions is entered I see
the Dial(SIP/100) on the console but the phone never rings...
I can pick up any extension and call 100 and it rings just fine and they
answer and ever
And it's wink-start on an E&M analog circuit, not on a standard analog
phone line from your telco. You would need a card that supports E&M to
do it even if the telco provided it (not sure if the Digium cards
support it, but I tend to doubt it).
Andrew Kohlsmith wrote:
On Thursday 13 October
On Thursday 13 October 2005 15:20, Apu Islam wrote:
> Is DID on analog line possible ? ( my telco is qwest) . Just wondering if
> there is any way to test it on anlog wcfxo cards.
Yes it's possible (winkstart can provide this) but No, Asterisk doesn't
currently support it on POTS lines and I'm fa
Andre Courchesne wrote:
This is not callerID but rather identify what tool-free number the
caller dialed in order to reach the PBX through the PRI line.
This is DNID, and should already be present in the appropriate channel
variable when the call arrives in Asterisk.
___
The SetCallerID command springs to mind. However, how do you know when
you are getting a toll free call? Does it come in to a certain extension
number on the PRI or is it just sent to the PRI's lead number?
If your CLEC sends you the call to a specific number and each of the 3
lines have a uni
Hi,
Ok, here is the setup. Asterisk conected to a PRI line (23 lines). 3
tool-free phone numbers are routed to this PRI line.
Customer wants to have a way to have shown on the receptionist phone
that the call comes from which of the 3 tool-free lines. Possibly
display on the phone that the
Well, Ishtiaq, to build on what Mark says...
First, I hate to be a grammar nazi, but you should use better grammar
in your emails. It looks very unprofessional using the word 'ur'.
Moving on, if this is a new install, which it appears to be, I would
do it *right* and put the solid investment in
I've done exactly this recently.
Frankly with hardphones being as cheap as they are I'd buy them. If you
are messing about with analogue adapters etc you'll end up with all
sorts of potentional echo problems not to mention the cost of the chanel
banks etc.
A hardphone will enable you many fe
hy all
actually i want to have a setup of five offices having round about 200 extensions ( each office having 35 to 45 ) which will be connected through asterisk.
now either i should go for voip phones( hard phones ). or use any interface card to asterisk server to which the analogue phones will be
Hi all,
Trying to build ztdummy on an old redhat 7.3 box running kernel
2.4.20-43.7.legacysmp. Yes, I have the kernel sources installed. Yes,
I set them up with make oldconfig; make dep.
The build error is:
make ztdummy
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__
-DEXPORT_S
I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes"
in the PAP2s.
However, there's sitll no callwaiting on the PAP2s. Everything else work
fine. Any ideas? Am I missing something somewhere?
Hi Andy,
You also need to set "CW Setting: Yes" on the User 1 and User 2 screens.
O
Hi,
I can't seem to find CW Act Code: and CW Per Call Act Code: in PAP2.
Does anyone know what they are in PAP2?
thanks.
AK
On 10/13/05, Tom Vile <[EMAIL PROTECTED]> wrote:
Have a look at the CW Act Code: CW Per Call Act Code: and remove the entries in there. I have a sipura so I dont know i
Have a look at the CW Act Code: CW Per Call Act Code: and remove the
entries in there. I have a sipura so I dont know if they are
using the same terminology buts it the same hardware.On 10/13/05, Andy Kuo <[EMAIL PROTECTED]> wrote:
Hi,
I have "callwaiting=yes" in my zapata.conf, and "Call Waitin
Hi,
I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes" in the PAP2s.
However, there's sitll no callwaiting on the PAP2s. Everything else work fine.
Any ideas? Am I missing something somewhere?
Thank you.
AK
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On Thu, 13 Oct 2005 08:41:17 -0400, Paul <[EMAIL PROTECTED]> wrote:
>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 intera
--- Blake Krone <[EMAIL PROTECTED]> wrote:
> What is the best solution? I dont want to have
> modify firewall's at all or
> do port fowarding. Ideally I would like a solution
> that with either a
> softphone or wireless hardphone one could connect
> via friends, family, or
> hotspots without reconf
Seems like my BYOD-Lite incoming has
been working without any interruptions...
I don't know about the outgoing; i
use different service for that.
rajesh
- Original Message -
From:
Thameem Ansari
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thu
I have a very odd problem. Normally incoming calls enter my dialplan
and person chooses an extension and I see this below:
-- Executing Dial("Zap/1-1", "SIP/102|20|") in new stack
-- Called 102
-- SIP/102-XXX ringing
For this installation I am NOT seeing the RING!
-- Executing Dial(
canuck15 wrote:
Nice!
Looks kinda like lego using cct boards from that angle ;).
Any idea what the echo canceller might cost?
No word on the pricing on the echo canceller yet, but I have put more
pictures of the boards up at http://www.telephonyware.com/sangoma which
should show the boards
Thanks to some help from this list, I can remap useless buttons (such as
services). However I can only remap to other things that are also
useless. So my question is:
Does anyone have more detail on how to use the functions listed in the
Admin Guide in section 4.6.1.15. I would like to be a
BJ Weschke has joined our bug marshal team and will be helping with code
reviews and other types of bug support... so please welcome him and give
him lots of work to do!
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Hi,
The VERY BEST
solution, is AIX... SIP + NAT tends to be lousy... For softphone you can try the
simplest one, DIAX, it is extremely basic, but does a good
job.
Now... If SIP
is a must, then it really depends a lot on the NAT you have there.. the best way
to do it is u
Matt,
thanks. do you know if the cheaper fxo cards will handle it ? this is basically for a test to check some billing engines.
I am wondering if anyone had done this in the past. any clue whether Qwest provides this kind of services on U.S. ?
-apu
On 10/13/05, Matthew Crocker <[EMAIL PROTEC
Yes, DID on analog line is possible. Your carrier may or may not
support it. For the most reliable service you'll want a 'wink
start' line with DMTF digits. With a DID line you supply
'battery' (-48VDC) to the phone company. They bring the tip to
ground for a second to signal an inc
I see a lot of comments but no actual show runs.
Can someone post a 7206 config.
I am having a dickens of a time getting calls to pass.
I currently have the following loaded.
Cisco IOS Software, 7200 Software (C7200-IK9O3S-M), Version
12.3(8)T6, RELEASE SOFTWARE (fc2)
Thanks !!!
Is DID on analog line possible ? ( my telco is qwest) . Just wondering if there is any way to test it on anlog wcfxo cards.
-apu
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
htt
Dear friends,
im very new to asterisk, even diz z my 1st mail to the list. am working
a small company & it has to main CDMA telepone connections. now
they wants to deploy a pbx & get out 20 nods(telephone lines) so is
it possible using Asterisk? & our both CDMA phones are HUAWEI
ETS2000 Series mod
Except DCA proxy server all other proxies timed out..
-ThameemOn 10/13/05, Marco Supino <[EMAIL PROTECTED]> wrote:
Yes, i am having timeouts on registering to the LAX sip server ofbroadvoice.Marco.Nate Kapi wrote:> I've been having a lot of problems with Broadvoice lately. Anyone else> been withou
Hi Steve,
On Thu, 13 Oct 2005, 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?
Bret explained mostly what the software does in a basic use case where you
would like a nice window
"Steve Gladden" <[EMAIL PROTECTED]> wrote:
>Am I still doing something wrong here?
I don't have any advice to offer, but I can sympathize. This is a
poorly documented area of Asterisk. I think it is quite a poor design
from a SIP point of view, although it may make some sense for a
monolithic P
On 10/14/05 00:57 Walt Reed said the following:
get 3 full length / height cards in a DL380. If they offered a single
card 12 / 16 port version (using 4 port modules,) they should be able to
hear, hear ! for 16-20 port type implementations, it is much more
economical to get 8-16 port FXO/FXS
Hi everyone
I’m using Asterisk as a transcoding gateway between
G711 and ilbc, and when I send out of band DTMF (rfc2833), the DTMF packet sent
has the same timestamp as the last RTP packet received by Asterisk. The problem
is that when I stop sending RTP, and only want to send DTMF pac
The file used with a Digium IAXy device: "iax.conf"
has the line: "port=5036" (I also use the "bindaddr=192.168.1.91" entry)
but when Asterisk talks to the IAXy device it
used port 4569 (from "tcpdump"). How are the port numbers assigned?
What tells the IAXy which port to use. The IAXy provisionin
On Thu, Oct 13, 2005 at 12:25:51AM +1000, Mark Lipscombe said:
> This is at http://www.telephonyware.com/sangoma
>
> In the mean time, here is some more information so this thread hasn't
> been a waste of time. The new cards will be available soon, and will
> also have an option for an addon 16
The ("/var/lib/asterisk/agi-bin/phpagi.php") is the newest form the site
updated today, and i wrote the script like other examples and i can't find a
syntax mistake inside extension.conf and the php script .(
On Thu, 13 Oct 2005 09:32:14 -0500
Moises Silva <[EMAIL PROTECTED]> wrote:
for some
Thank you for the reply. All of the serves are running 1.0.9.
If jitterbuffer and the like are not available, why have those options
available from the non-CVS/HEAD release and in a series that does not
support such features. I don't seem to recall reading that anywhere else -
not an argument agai
I'm hosting some folks today from Sangoma for a sales training. If
anyone on the list has any specific questions regarding current or
upcoming Sangoma products, please shoot me an email in the next hour or
so and I'll do my best to get you the answer.
Thanks
--
Cory J Andrews
Partner / Purch
Nice!
Looks kinda like lego using cct boards from that angle ;).
Any idea what the echo canceller might cost?
> -Original Message-
> From: Mark Lipscombe [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, October 12, 2005 7:26 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
We are Voip distributors
in Europe so we can buy them from Manufacturer
cheaper.
Anders
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francis Ballares (VoIPware.ca)
Sent: den 13 oktober 2005 17:20
To: Asterisk
Users Mailing List - Non-Commercial Discussion
FYI, this is a bug that has been patched in cvs-head - see
http://bugs.digium.com/view.php?id=4468>
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gary Reuter
Sent: Thursday, October 13, 2005 9:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discuss
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
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
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 w
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 A
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] PA168S/AT32
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
ac
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
problems
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 Be
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 o
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 - Non-Commercial
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
>
> __
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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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
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
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.h
I just tested it and it's working fine.
Does your Linux box have internet access?
--
Cheers,
Matt Riddell
___
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http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
__
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), I’m
ready to install spandsp, app_txfax & app_rx fax to try fax to email &
email to fax.
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 fir
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 se
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 * c
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 Gmb
I’m wanting both the voice and the
configuration to be secure. (very secure). I don’t care if it is SIP or
IAX but I do need a softphone on the pocketpc I can use. I’d appreciate
if you could take a look this weekend for me.
Thanks, -Peter
From:
[EMAIL PROTECTED]
[mailto
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
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http
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 mail
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 D
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) an
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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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, 2
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
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 fxo
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]
type=friend
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: [E
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)
506-854-
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
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
fa
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
ad
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 ;)
---
Marek
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
_
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:
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 mu
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
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
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
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", "4989427xx
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"
Sent: Thursday, October 13, 2005 7:58 AM
Subject:
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
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
as
> 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.
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Asterisk-User
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 Se
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
As
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
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