I am sending faxes from a standalone fax and going through a
TDM400 ( 2xFXS, 2xFXO ). I dont want to send over the internet, in fact
the only reason I am going through the card at all is to capture the dialing
for call accounting.
Everybody tells me to use G.711 but I dont see how
you
Hi,
I am new with asterisk and everything that deals with. I was wondering if there is a way to call a OH323 user or SIP user using Netmeeting/SJPhone with H323 as the default protocolwithout having a gatekeeper.
I can make a call from SIP to OH323 by specifying it in the extensions.conf file,
*CLI show version
Asterisk CVS-HEAD-03/13/05-23:38:12 built by [EMAIL PROTECTED] on a x86_64 running
Linux
How can I upgrade safe?
How can I downgrade if something did not work out right?
What should I upgrade?
Where can I read for each package the changes to see if it is worth to
upgrade?
Is
Hello all,
I'm looking for a free SIP softphone with IM feature supporting Jabber
and Yahoo! Messenger protocol. The OS of choice is WinXP/Win2k.
Any help appreciated.
Regards,
Thai
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I am lucky that everytime I want to lookup something on the wiki, it is
not available, ... Last time I was lucky to read about faxes, but was
more or less confused about many different approaches to solve it.
What can I get ? (Which feature, which comfort, fax in, fax out, )
I have
I got this from the voip wiki but the original script didn't seem to
work right so I fiddled with it a little bit. I am no expert so maybe
someone can look at it for errors. This is for my cable connection. So
far asterisk seems to use 1:10 while all other traffic uses 1:102. How
does one
Hello Sir,
I use FXO card but cann't get CallerID.
Here is Taiwan andthe callered follows ETSI in permitting DTMF and FSK signals
(Seems same to that in Australia and New Zealand, based on Bellcore)
Does anybody know what's the cadence setting will be in zapata.conf, so that I can get the
It would help if you would include your config files and state what type
of FXO device you have
On Sun, 10 Apr 2005, Dominic Lu wrote:
Hello Sir,
I use FXO card but cann't get CallerID.
Here is Taiwan and the callered follows ETSI in permitting DTMF and FSK signals
(Seems same to that in
Rich Adamson ...
Okay. There are two fairly small integrated circuits on the board.
Can you post the part numbers on those chips?
Not at the moment, unfortunately. The card is in a datacentre 200 miles
away. However, and while I appreciate that you can't read anything from
this, but this is the
Release notes:
Version 0.77 - 10. April 2005.
* All strings can now be translated - new strings file published on forum
* Optimizations of code
* Bug fixes
Download here: http://ipswitchboard.thorben.dk
Would you like to see IPSwitchBoard in your own language?
Hi, All
I installed a PortaOne's Radius client for my asterisk
Server. But I can't run ast-rad-acc.pl after installation. It says "Can't call
method "val" on an undefined value at ./ast-rad-acc.pl line 293." It also show
"Config file error" in the log file.
Has any one meet this
Hi,
I've been having some echo problems on SIP calls with 1.0.7, so I
wanted to take the latest CVS to see if it has the same issue. However,
I cannot get it to compile and having searched these lists and Googled
elsewhere, I can't find any references to a similar problem. This is
what I get:
I have two snom phones, one is a Snom 190 on my desk and one is a soft
phone snom 360
Extension 615 works fine, but all connections with 616 works not.
On 616 you hear the other party, but the other party get only a white
noise sound !! This is indipendent which direction I call.
What might be
Asterisk is on my box now running about one month without any troubles.
Since two days I got troubles:
1. The Zapta card (2 FXS, 2 FXO) suddenly does not like one phone. It
simple does not supply with a dial tone. You cannot dial. You can reach
it, better say, you can dial it, it rings, but no
cmisip wrote:
far asterisk seems to use 1:10 while all other traffic uses 1:102. How
does one packet shape RTP?
Thanks for any help.
# +-+
# | root 1: |
# +-+
# |
# ++
# | class 1:1 |
#
Hi List,
I'm glad to announce Asterisk::LCR 0.02, a free/GPL, simple, modular
least cost routing AGI engine for asterisk. It's written in Object
Oriented Perl and aims at being extensible, documented and well tested.
Features:
- Imports rates from NuFone, VoIPJet, LiveVoIP (more providers can
Howdie folks,
Is it possible to play a different dialtone as soon as a user dials say
'0' for an outside line ? Ignorepat is an inadequate solution because
local users are accustomed to getting a specific PSTN dialtone. I need
an audible change in the frequency/modulation of the tone.
Thanks,
Hi all, I am trying to set up two asterisk servers (SrvA and SrvB),
and what I want to get done is that if I dial 1X on SrvB the call must
be routed to extension X on SrvA and if I dial 2X on SrvA the call
must be routed to extension X on SrvB. I've read the www.voip-info.org
wiki abouta
[DC]
Well mine is legitimate digium
And I'm in the usa
Here is the output but I have no idea what that means?
[EMAIL PROTECTED] root]# cat /proc/interrupts
CPU0
0: 490763 XT-PIC timer
1: 2 XT-PIC keyboard
Okay. There are two fairly small integrated circuits on the board.
Can you post the part numbers on those chips?
Not at the moment, unfortunately. The card is in a datacentre 200 miles
away. However, and while I appreciate that you can't read anything from
this, but this is the card:
Hi all, I am trying to set up two asterisk servers (SrvA and SrvB),
and what I want to get done is that if I dial 1X on SrvB the call must
be routed to extension X on SrvA and if I dial 2X on SrvA the call
must be routed to extension X on SrvB. I've read the www.voip-info.org
wiki
Hi
Is there a search engine for this mailing list ?
Thanks
Patrick
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Patrick,
The way I always do it is through Google. You can search for a specific
term by entering the term followed by site:lists.digium.com (not
including the quotation marks).
DS
On 10 Apr 2005, at 15:43, Equipe du Royaume wrote:
Hi
Is there a search engine for this mailing list ?
Thanks
Is there a search engine for this mailing list ?
Google for:
site:lists.digium.com search terms
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
___
Asterisk-Users mailing list
Is there a search engine for this mailing list ?
Yes, and it's the best search engine in the world.
Just go to google and type this :
site:lists.digium.com somesearchstring
hth
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Apr 10, 2005 8:29 AM, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
Hi List,
I'm glad to announce Asterisk::LCR 0.02, a free/GPL, simple, modular
least cost routing AGI engine for asterisk. It's written in Object
Oriented Perl and aims at being extensible, documented and well tested.
I've
I am currently having a problem with asterisk 1.0.7 where after exactly 10
minutes of a call the inbound audio stops. My setup is fairly simple. I have
a FreeBSD box running asterisk connected to the internet. It has a SIP or
IAX client talking to it on a private network. The BSD box talks SIP
hello people,
I'm using the Asterisk arithmetic operations to do multiplications and
divisions. However, they just do integer calculations whereas i need
to do float point calculations.
Does anyone know any way I could do them ?
Please repond asap
___
*CLI show version
Asterisk CVS-HEAD-03/13/05-23:38:12 built by [EMAIL PROTECTED] on a x86_64
running
Linux
How can I upgrade safe?
How can I downgrade if something did not work out right?
What should I upgrade?
Where can I read for each package the changes to see if it is worth to
On April 10, 2005 10:50 am, Rich Adamson wrote:
One way to do that is simply:
[ snippage of simply mv'ing to a backup ]
That is *precisely* how I do it for small changes, and for full-out upgrades,
I have the old slackware packages standing by.
-A.
I initially used that script without modification. However, I noticed
that all traffic was going through class 1:102 regardless. Seems as if
all the children of 1:20 are set with a prio of 0 by default even if
1:20 is specifically set to prio of 2. I used
/sbin/tc -s -d class show dev eth0
to
Exact same situation with analogue phone happened to me. Apparently,
there are still some serious bugs in zaptel (possibly in firmware of
Digium cards, even). As far as analogue phones are concerned, my
experience is that IAXy is much more reliable than TDM cards.
Ronald Wiplinger wrote:
Hi All,
I'm just starting with Asterisk, so this may be something very
simple. I'm using a X100P, and a softphone (GnomeMeeting) with
OH323 providing the linkage into Asterisk. The (very) simple
setup looks like this:
PSTN---X100PAsterisk---OH323 GateKeeper---GnomeMeeting
(zap channel)
This depends on what kind of phone you are using.
With most (any?) SIP phones, nothing will be sent by the phone to the
server until it actually dials (whereas Skinny phones sent out on/off
hook and digits realtime).
If you're using a Cisco phone with a sip image, my guess is that you
can set
On April 10, 2005 04:47 am, cmisip wrote:
I got this from the voip wiki but the original script didn't seem to
work right so I fiddled with it a little bit. I am no expert so maybe
someone can look at it for errors. This is for my cable connection. So
far asterisk seems to use 1:10 while
How Can i make conferance like this
Call came from PRI
And joining Called Number Conferance Room (211)
While joining progress.
I want to make Asterisk call sip agent for 2nd conferance person
When sip agent answer then SIP agent join to room(211)
1st.Conferance Person (PRI)
2nd.
Jean-Michel Hiver wrote:
Hi List,
I'm glad to announce Asterisk::LCR 0.02, a free/GPL, simple, modular
least cost routing AGI engine for asterisk. It's written in Object
Oriented Perl and aims at being extensible, documented and well tested.
DEPENDENCIES
This module requires these
Rich Adamson wrote:
snip
The TDM400 card (with appropriate modules) is the replacement for the x100p, and the chipset in use on those modules _do_ support many different international standards. The early versions of the TDM card had several problems (output of dmesg showed Wildcard TDM400P REV
On April 9, 2005 08:25 pm, Eric Wieling wrote:
Which specific Digium card does not use the TigerJet chip (as shown in
lspci)?
TE405P:
05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev
01)
I imagine the TE410 and TE110 are both also similarly lspci'd.
I
In order for this to be helpful, you need to recompile with make valgrind
and edit your Makefile and turn on all the debugging stuff.
-Matthew
From: Paul P. Pongco [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion
This has already been answered...but I can't find it...
Has anyone set up multiple fax lines in asterisk...
Fax Extension #1 goes to email1
Fax Extension #2 goes to email2
ETC...
In other words, I want to be able to give numerous users each
a virtual fax machine..
Bill
cmisip wrote:
I initially used that script without modification. However, I noticed
that all traffic was going through class 1:102 regardless. Seems as if
all the children of 1:20 are set with a prio of 0 by default even if
1:20 is specifically set to prio of 2. I used
My setup is a
Jim,
Thanks for sharing this. I am currently using cidlookup.agi written by
James Golovich. http://asterisk.gnuinter.net/
However the problem I have with that script and probably this one also
is that my provider sends the number as +16105551212 so I need a way
to strip out the leading two
I'm sorry, I don't mean to knock on digium but this quote at a highly
competitive price. is just wrong, and for some reason, it struck a nerve.
I guess they are saying this because its IAX and not SIP? Its not
competitive because I can buy a PAP2-NA for $50. Granted, the PAP2s don't
speak IAX,
Hi,
I am running 1.0.6 STABLE and I use SMS occasionally, but not often. I
wanted to send one today and discovered that the SMS app would still
receive and process them, but smsq doesn't seem to be formatting the
call files properly.
What is missing in the .call file is the destination number and
Rich Adamson - would appreciate your advice as well, as your mail is the
closest I have seen to a knowledgeable response so far in regards to
this crackling issue. I have a customer who has a very similar
crackling problem and to date we have suspected it to be the ISDN BRI
adapter and/or
The TDM400 card (with appropriate modules) is the replacement for the x100p,
and the chipset
in use on those modules _do_ support many different international standards.
The early versions
of the TDM card had several problems (output of dmesg showed Wildcard TDM400P
REV E/F as an
example),
After some sleep and a good breakfast, I seem to be able to think more
clearly.
I have come upon these conclusions:
1. Qos is all about managing upload packets ( and download packets
indirectly by managing upload packets).
2. The ceiling kbit actually refers to your upload speed. It is
Hi all,
I am
trying to use Unicall stack in Indian
environment. I got the
calls into the system but asterisk is not picking up the call and when we debug
on the channel it says Protocol variant null even after specifying the variant.
please help me out in this .
Apr
9 01:10:57
It seems Petal.pm (maybe a part of FreezeThaw???) is missing on my
system. Where can I find it?
It's a 'cut and paste' from another script, and it's a mistake. I'll
remove the dependency.
Meanwhile, you can 'fix' the issue by installing the module:
perl -MCPAN -e 'install Petal'
Alternatively,
Well I am doing this with success. Dont now that its
necessarily the right waywhat I did was create an NFS
share on the machine that I wanted to store all of the voicemail onthen
changed asterisk.conf spool directory to the appropriate /mnt drivemove
all the folders that were in
Jean-Michel Hiver wrote:
It seems Petal.pm (maybe a part of FreezeThaw???) is missing on my
system. Where can I find it?
It's a 'cut and paste' from another script, and it's a mistake. I'll
remove the dependency.
Meanwhile, you can 'fix' the issue by installing the module:
perl -MCPAN -e
Version 0.78 of IPSwitchBoard has just been released.
* Records all incoming and outgoing calls on separate tab page
* Make calls by double clicking the incoming/outgoing call
* Italian language updated - thanks to Francesco Romano
FREE Download: http://ipswitchboard.thorben.dk
Would you like
More known for their line of analog terminal adapters (ATA's), Sipura has
released their first business hard phone. Small in size and large in features,
The Sipura SPA-841 IP telephone can be configured as a two (2) line or, via a
simple software upgrade, a four (4) line full featured
Here is my problem
I have an incoming call on the FXO port of the Sipura 3000 this goes to
my asterisk box (running CVS - head) and then by default goes back to
the sipura 3000 on the fxs tel port.
Every 5 or so calls when the call is picked up on the fxs port telephone
I just get beeping and
Rich Adamson - would appreciate your advice as well, as your mail is the
closest I have seen to a knowledgeable response so far in regards to
this crackling issue. I have a customer who has a very similar
crackling problem and to date we have suspected it to be the ISDN BRI
adapter
Here is my problem
I have an incoming call on the FXO port of the Sipura 3000 this goes to
my asterisk box (running CVS - head) and then by default goes back to
the sipura 3000 on the fxs tel port.
Every 5 or so calls when the call is picked up on the fxs port telephone
I just get
Kerry Garrison wrote:
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=24
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=24
-Kerry
Just make sure you don't have a cordless or cell phone near by or the
headset jack will receive a considerable amount of
Trevor Peirce wrote:
Kerry Garrison wrote:
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=24
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=24
-Kerry
Just make sure you don't have a cordless or cell phone near by or the
headset jack will receive a
I appreciate everyone's help with setting up an external extension.
Here's a diagram
{SPA2000} - NAT1 - Internet - NAT2 - Asterisk - SIPPhone
SIPPhone is on the same internal subnet as *
NAT2 has a public/staic IP and ports are forwarded to Asterisk
I can successfully do the following:
1.
I didn't experience those issues. I use both 900mhz and 2.4ghz phones as
well as several cell phones in this location. Are you using the most current
firmware?
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor Peirce
Sent: Sunday, April 10,
Hello
I got problem to run Asterisk on virtual IP. I have rtpproxy installed.
Audio stream seems not able to sound back. When I call 'echo' extension,
asterisk starts 'echo'; but there is no echos.
It works well on physical IP.
Can someone help?
thanks!
steven
Date: Sun, 10 Apr 2005 11:06:59 -0500
From: Bill Ford [EMAIL PROTECTED]
Subject: [Asterisk-Users] Fax to Email
This has already been answered...but I can't find it...
Has anyone set up multiple fax lines in asterisk...
Fax Extension #1 goes to email1
Fax Extension #2 goes to email2
Yes, I used to see this problem from time to time as well. I'm not 100%
sure, but I think it was caused by someone dialing in to my Asterisk
box, then hanging up at the menu and the SPA3K not detecting the hangup,
or at least not sending it to Asterisk.
But when either another call comes
Hi everybody,
I'm new in *, i have installed over fedora core 3,
with kernel version 2.6 and ztdummy.
I have created one conference in meetme.conf and I
have modified properly extension.conf. But when I try to do a call to this
extension I get the following errors:
-- Executing
How about MWI? did you get it to work?
I was about to offer the same solution, using NFS :)
On Apr 10, 2005 2:01 PM, Jason Brown [EMAIL PROTECTED] wrote:
Well I am doing this with success. Don't now that it's necessarily the
right waywhat I did was create an NFS share on the machine
You can shape incoming (TCP) traffic by dropping packets that exceed
your download limit... But for this you rely on the other end to
actually decrease their sending speed, i.e., if they knowingly flood
you, there's nothing to do in your end..
The way to suggest to limit download speed by
You need to compile zaptel with ztdummy module.
Uncomment ztdummy in Makefile.
You need have to loaded this module as kernel module before executing meetme.
Here you will find more detailed info:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
Adam
Cytowanie Xisco
Hi,
Has anyone experienced any problem with polycom phones not sending voice?
Occasionally some phones didn't send voice, a few times phone didn't play
the incoming voice. It didn't happen often on the IP 500 I bought last
November. But it happened quite a lot on the IP 600 I got recently.
Trying to get my asterisk system to detect and transfer inbound fax calls.
System is supposed to detect the fax, and then ring my fax machine which is
connected to a Sipura device.
Here's my extensions.conf:
exten = 1000,1,Goto(s,1)
exten = 630446,1,Goto(s,3)
exten =
On Sunday 10 April 2005 5:15 am, Ronald Wiplinger wrote:
I have two snom phones, one is a Snom 190 on my desk and one is a soft
phone snom 360
Extension 615 works fine, but all connections with 616 works not.
On 616 you hear the other party, but the other party get only a white
noise sound
I just installed the latest version .8 of * @ home and it looks like it
installed successfully, however when i goto the box's IP address and
click on link to access the management portal the default password
(admin/password) do not work...anyone else have this issue?
thanks
bernie
My wife is from Russia, and we have a lot of friends here in the US
that have families back in Russia. While calling Russia is fairly
inexpensive, for someone in Russia to be able to call the US is a
different matter. So I'm looking at the best ways to setup a
callback system. Here is what I
Does anyone have a working TE110P that is
willing to share their /etc/zaptel.conf file ? Also, any suggestions or link to
troubleshooting this particular card?
Thanks
Tim
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Sun, 2005-04-10 at 16:21, Bernie Courtney wrote:
I just installed the latest version .8 of * @ home and it looks like it
installed successfully, however when i goto the box's IP address and
click on link to access the management portal the default password
(admin/password) do not
Message: 16
Date: Sun, 10 Apr 2005 12:45:23 -0700
From: Kerry Garrison [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Sipura SPA-841 Phone Review
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type:
On Apr 10, 2005 2:21 PM, snacktime [EMAIL PROTECTED] wrote:
My wife is from Russia, and we have a lot of friends here in the US
that have families back in Russia. While calling Russia is fairly
inexpensive, for someone in Russia to be able to call the US is a
different matter. So I'm
maint/password
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bernie
Courtney
Sent: Sunday, April 10, 2005 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk @ Home AMP problem
I just installed the
I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on
Fedora Core 3.
Firstly, despite the warnings in h323/README, I decided to try using
the distro-specific versions of openh323 and pwlib. Of course, the
Makefiles in channels and channels/h323 assume that openh323 and pwlib
have
On Apr 10, 2005 2:21 PM, snacktime [EMAIL PROTECTED] wrote:
My wife is from Russia, and we have a lot of friends here in the US
that have families back in Russia. While calling Russia is fairly
inexpensive, for someone in Russia to be able to call the US is a
different matter. So I'm
Does anyone have station monitoring working on the Snom 360 softphone?
I have Snom 360 softphone ext 360 and I want to monitor Cisco 7940 ext 301.
How do I configure my extensions.conf? I've tried going by the wiki but it
just doesn't seem to work.
The snomSoft-SIP 3.60a is still beta quality and known to have some
audio problems on some systems,
I believe that this will improve soon.
If you have multiple audio devices on the PC you might want to try
different combinations.
Ronald Wiplinger wrote:
I have two snom phones, one is a Snom
On Apr 10, 2005 3:52 PM, Rich Adamson [EMAIL PROTECTED] wrote:
On Apr 10, 2005 2:21 PM, snacktime [EMAIL PROTECTED] wrote:
My wife is from Russia, and we have a lot of friends here in the US
that have families back in Russia. While calling Russia is fairly
inexpensive, for someone in
Actually, these problematics are classical to many countries:
1-) If you put russian sim card in a GSM modem, it can receive missed
calls from people who call and hang up and their caller id (most local
mobile operators generally get caller Id). Based on this input from GSM
modem, you can
On Apr 10, 2005 3:17 PM, Hakem Taourchi [EMAIL PROTECTED] wrote:
Actually, these problematics are classical to many countries:
1-) If you put russian sim card in a GSM modem, it can receive missed
calls from people who call and hang up and their caller id (most local
mobile operators
Thanks...
I suppose also if I have spanDSP on the * box, I'd modify the config
lines accordingingly?
Bill
On Apr 10, 2005 2:57 PM, Justin Newman [EMAIL PROTECTED] wrote:
Date: Sun, 10 Apr 2005 11:06:59 -0500
From: Bill Ford [EMAIL PROTECTED]
Subject: [Asterisk-Users] Fax to Email
This
Is there anyway to append the '#' symbol to a dial string? - hex/octal
whatever? I'm surprised that I can't find anything searching the wiki or
google.
Thanx
John Breedn
Hawaii
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hello list,
I keep having repeated crashes with * one or two times a week
I'm using Asterisk CVS-HEAD-04/02/05-22:38:19,
Asterisk is run with -vvvr
Apr 11 09:02:01 NOTICE[15659] chan_sip.c:-- Registration for
'[EMAIL PROTECTED]@x' timed out, trying again
Apr 11
On April 10, 2005 05:03 pm, John Novack wrote:
As to that hold button. What idiot decided it should be in the middle
of a row of keys, the same size as the others, and not a bright color?
Maybe me; I have no desire for a bright 'hold' button. Give me the Norstar
system where 'Rls' (release,
On April 10, 2005 12:01 pm, Matthew Boehm wrote:
I have a TE405P and mine shows up as Xilinx but a lvl 2 tech a digium says
it still uses the TigerJet chipset. That's why it won't work in my Dell.
I'll paypal you US$100 if you can find a TJ320 chip on either the TE410P or
TE405P. It doesn't
I am using [EMAIL PROTECTED] 0.8 with a single X100P clone card. I am having
trouble making outbound calls on certain lines. I have three analog lines at
home. Line 1 has caller ID, 2 3 do not. I can always call in on any of the
three lines, but I can make only one outbound call on 2 or 3 after a
I have a Sipura SPA-841. I have configured my SIP.conf and extensions.conf
to allow the phone to dial an outside number via my Zap/2 PSTN. When I pick
up the handset I get a dialtone, however, when I press 9, the dialtone
stops. I assumed it would pause for a moment and give me another dialtone
On incoming SIP calls, the caller just gets silence instead of ringing
until * answers the channel. Is this a configuration issue on my
end?
Chris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Apr 10, 2005 5:31 PM, snacktime [EMAIL PROTECTED] wrote:
On incoming SIP calls, the caller just gets silence instead of ringing
until * answers the channel. Is this a configuration issue on my
end?
Chris
Correction, this is true for both IAX and SIP incoming calls on my
system. I
On Apr 10, 2005 5:28 PM, Paul [EMAIL PROTECTED] wrote:
I have a Sipura SPA-841. I have configured my SIP.conf and extensions.conf
to allow the phone to dial an outside number via my Zap/2 PSTN. When I pick
up the handset I get a dialtone, however, when I press 9, the dialtone
stops. I assumed
You could record the sound of a dialtone and background(dialtone) while the
dialplan is waiting for the rest of the number...
Exten = _9.,1,background(dialtone)
Exten = _9NXXNXX,2,dial(...etc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Never mind I had a dumb typo.
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, April 10, 2005 5:12 PM
Subject: [Asterisk-Users] snom360 hint priority
Does anyone have
That should be working ok.
Check the internal web page of the phone and look at the sip trace to
see if the phone is getting the NOTIFY messages.
dialplan should have:
exten = 301,hint,SIP/360
Henry Devito wrote:
Does anyone have station monitoring working on the Snom 360 softphone?
I have Snom
On Apr 10, 2005 5:33 PM, snacktime [EMAIL PROTECTED] wrote:
On Apr 10, 2005 5:31 PM, snacktime [EMAIL PROTECTED] wrote:
On incoming SIP calls, the caller just gets silence instead of ringing
until * answers the channel. Is this a configuration issue on my
end?
Chris
Correction,
On Apr 10, 2005 5:31 PM, snacktime [EMAIL PROTECTED] wrote:
On incoming SIP calls, the caller just gets silence instead of ringing
until * answers the channel. Is this a configuration issue on my
end?
Chris
Correction, this is true for both IAX and SIP incoming calls on my
1 - 100 of 130 matches
Mail list logo
