Michael Ulitskiy wrote:
Hi,
I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I have H.323 to PSTN
gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI.
Everything works fine with one exception. I seem to be unable to figure out why I cannot hear
PSTN
SloopJohnB wrote:
Does anyone on this list know of any Linux based apps that will work with
Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It
seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their
firmware for their proprietary voice mails.
My wish list
Hachison wrote:
Hello all
I would like change the all announcemennt(Voicemailmain,Voicemail etc.).
But I don't know how to change the these each prompt.
Do we have any guide book for this?
Please teach me about changing the voicemail or other prompt.
You can change some prompts from a phone if
Hello all,
I wonder to have some feedback on using ISDN BRI Cards with Asterisk and
the Echo problem.
I have tried a simple BRI card with i4l driver and encounter huge echo
problem. I have tried to solve it with a Sw chocanceller without
success. What I'd like to know is wether some of you
Thanks for the info.
I call the Telco again and open an tracking failure of international dial.
And try again pridialplan=unknown and dial 00xx and now works.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Steve
Underwood
Enviado el: jueves, 20 de
Daniel ANDRE wrote:
Hello all,
I wonder to have some feedback on using ISDN BRI Cards with Asterisk
and the Echo problem.
I have tried a simple BRI card with i4l driver and encounter huge echo
problem. I have tried to solve it with a Sw chocanceller without
success. What I'd like to know is
For production environments we ONLY use the Eicon Diva server card
range, which supports on-board echo cancellation. However it is rather
expensive.
Tan
www.voiptalk.org
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: 21 November 2003
WipeOut a écrit:
Daniel ANDRE wrote:
Hello all,
I wonder to have some feedback on using ISDN BRI Cards with Asterisk
and the Echo problem.
I have tried a simple BRI card with i4l driver and encounter huge
echo problem. I have tried to solve it with a Sw chocanceller without
success. What
Hi,
Try this one:
Dual Tone Multifrequency Relay for SIP calls Using Named Telephone Events
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guide09186a0080087edb.html
Andrey.
This is a password-protected document (CCO account required.) Can
you refer to a
Hello
I can cange busy and unavaile message by voicemailmain.
I would like to change the Voicemail and Voicemailmain prompt itself.
Hachison wrote:
Hello all
I would like change the all announcemennt(Voicemailmain,Voicemail etc.).
But I don't know how to change the these each prompt.
Hi all,
I sent this question to Digium support as a sales enquiry three days ago,
but I thought I'd try here too in the meantime:
We currently run a telecom switch in Brussels, Belgium, which uses 4
Dialogic Quad cards totaling 16 E1s running on a win2000 server.
We are simply re-routing calls,
On 21/11/03 10:06, Richard Bennett wrote:
Are you aware of any motherboards with 4 x 3.3 volt PCI slots, or will there
be a 5 volt version of the card available soon?
AFAIK, Digium is testing the TE405P (or whatever they're going to call
it) right now. There seem to have been some delays - it
Hi,
i did some tests. If the kernel modules are loaded
Module Size Used byNot tainted
wct1xxp11488 31
zaptel176256 64 [wct1xxp]
ppp_generic19068 0 [zaptel]
slhc5392 0 [ppp_generic]
and the E1 is
At 15:38 -0500 20/11/03, Billy Huddleston wrote:
Use CIPE, It's a UDP based VPN solution.
Don't use CIPE, it has holes in it and is breakable.
Use IPsec either FreeS/WAN or if you are running 2.6 kernel you can use its
IPsec stack. Or an appliance to provide tunnels if you don't want to fiddle
Richard Bennett wrote:
Are you aware of any motherboards with 4 x 3.3 volt PCI slots, or will there
be a 5 volt version of the card available soon?
Do you really want all those spans going down cause someone tripped over
a power cable or your hard drive nukes itself?
We only use 2 TE410Ps in
hi,
i am having callerid problems with *. i have the
callerid from my telco and it shows up in my normal
phone when i connect it directly to the line but if i
connect the same phone thru * server the callerid is
not shown. i am using X101p and tdm400p. i have
everything defined in my zapata.conf
Tilghman Lesher wrote:
On Thursday 20 November 2003 17:15, Adam Hart wrote:
Reinhard Max wrote:
Asterisk uses UDP, but ssh can only forward TCP ports.
Ahhh something I completly missed...that makes sense because I
tunnel lots of other things...Are there other protocols that are
TCP instead?
All
On Wed, 19 Nov 2003, Florian Overkamp wrote:
[...]
add a link to the FAQ on the bottom of every message, much like the
subscribe/unsubscribe instructions. Then again, nobody seems to read those
either :-P...
Not sure about what mailing list footer you get, but I do only get this at
the bottom
Hi Hachison.
you need voice actor and sampler software and codec conversions.
mack_jpn
On Fri, 21 Nov 2003 19:00:07 +0900
Hachison [EMAIL PROTECTED] wrote:
Hello
I can cange busy and unavaile message by voicemailmain.
I would like to change the Voicemail and Voicemailmain prompt itself.
On Thu, 20 Nov 2003, marrandy wrote:
On Monday 17 November 2003 10:31 pm, Brian West wrote:
Show us your sip.conf entries.. and i'm sure I can point out the error.
bkw
Well, I've tried over 20 different settings, from examples in the archives
etc.
This is the last one I tried, for
Senad Jordanovic wrote:
Does this implementation of H323 for * terminates and originates calls
successfully from Cisco 5300?
Can't tell. I don't have a 5300 to test it.
Michael.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I'm testing the Eicon Server BRI/PRI cards for the moment and they are very
satisfactory because they have some interesting features onboard like echo cancelation
() and onboard encoding see www.eicon.com
Van: Daniel ANDRE [mailto:[EMAIL PROTECTED]
Senad Jordanovic wrote:
Does this implementation of H323 for * terminates and originates calls
successfully from Cisco 5300?
The H.323 driver that ships with Asterisk most certainly inter-operates
with As5300.
Jeremy McNamara
___
Asterisk-Users
Il ven, 2003-11-21 alle 11:29, Fearghas McKay ha scritto:
At 15:38 -0500 20/11/03, Billy Huddleston wrote:
Use CIPE, It's a UDP based VPN solution.
Don't use CIPE, it has holes in it and is breakable.
I've started testing OpenVPN and it doesn't seem to be that bad. Its
main weakness, IMHO,
I'm trying to have the following small architecture:
PSTN
|
Linux box with an public IP and Asterisk
|
Internet
|
Firewall/NAT/Router
|
Fearghas McKay wrote:
Don't use CIPE, it has holes in it and is breakable.
Use IPsec either FreeS/WAN or if you are running 2.6 kernel you can use its
IPsec stack. Or an appliance to provide tunnels if you don't want to fiddle
with kernels.
I have FreeS/WAN setup for my wireless...I guess its
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
What does Asterisk SIP support?
What I mean is, is it just for VoIP calls or i can send
message to someone?
As i understand with SIP i can contact [EMAIL PROTECTED]
and communicate via voice, video or whatever.
Can i use Astersik for anything else
C M wrote:
hi,
i can get callerid in my phone directly connected to
the pstn line. when i cannoect it o * it doen't give
me callerid. i have set usecallerid=yes in zapata.conf
file/
whaat could have happened?
Has it ever worked? I had soemthing similar with a new FXO card...as it
turns out
Hello,
My Cisco phone has software:
Boot Load: PC030300
Ver: 3.2(7.0)
And I want to upgrade it to SIP 6.0
Is it possible or I have to upgrade to ealier then
6.0 and then to 6.0 ?
bart
not with *.
i guess its the country issue. like i saw som posts
with callerid issues in uk. i am in nepal. how can i
configure * AGI to make it compatible with my country.
has anyone written a script?
cm
--- Chris Hirsch [EMAIL PROTECTED] wrote:
C M wrote:
hi,
i can get callerid in my
Jeremy McNamara said:
Do you really want all those spans going down cause someone tripped
over
a power cable or your hard drive nukes itself?
How's this worse than an as5300? I could install ata-flash and get
high-ish end pc hardware (rcc serverworks boards, etc). Heck, if it
was such an evil
We have 8 z-plex and they are fairly new! We replaced them with what is the
best so far. adtran 750. We got them on ebay for around $ 500.00 new! With
all the 24 fxs ports!
- Original Message -
From: Alex Pavlovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, November 20,
Anton L. Kapela wrote:
Jeremy McNamara said:
Do you really want all those spans going down cause someone tripped
over
a power cable or your hard drive nukes itself?
How's this worse than an as5300? I could install ata-flash and get
high-ish end pc hardware (rcc serverworks boards, etc).
I am trying to use the SAY NUMBER command from an AGI script but it does
not seem to be working..
If I use EXEC SayNumber 2 and execute the asterisk command from the
AGI it works and I hear the 2 said on the phone..
If I use SAY NUMBER 2 I see -- Playing 'digits/2' (language 'en') on
the
On Fri, 2003-11-21 at 07:14, C M wrote:
not with *.
i guess its the country issue. like i saw som posts
with callerid issues in uk. i am in nepal. how can i
configure * AGI to make it compatible with my country.
has anyone written a script?
AGI is not configurable, and also would not help
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Lawson
Sent: Thursday, November 20, 2003 7:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Solved! Snom 200 Busy signal
As a follow up to my earlier posting, the problem with the Snom 200
Busy
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Fri, 2003-11-21 at 07:14, C M wrote:
not with *.
i guess its the country issue. like i saw som
posts
with callerid issues in uk. i am in nepal. how can
i
configure * AGI to make it compatible with my
country.
has anyone written a
On Fri, 2003-11-21 at 08:17, C M wrote:
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Fri, 2003-11-21 at 07:14, C M wrote:
not with *.
i guess its the country issue. like i saw som
posts
with callerid issues in uk. i am in nepal. how can
i
configure * AGI to make it
Mark-
The system is Tyan s2723, which includes an integrated graphics controller,
and I've connected a monitor to this port. I don't start the X-windows
software however, and I don't use this monitor. For starting my scripts, I
connect using SSH via an Ethernet port, however everything operates
James Sizemore wrote:
I did not even know about it! But seeing as it is not in the change
log no wonder?
You have the bug number the notes are under for usage?
ID # 345
10/02/03 - logger_reload.diff
Summary - 'logger reload' CLI command
Description - Closes and reopens the log files.
Title: Message
Yes,
it is. But why would you want to do that when yo said what you want it to
be at 6.0.
Maybe
you didn't expling what and why you want to do it in enough detail to get a good
answer.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Thanks Dorian for the response. Guess you are using Gentoo as any other
flavor on fully loaded mode.
Though its nice to know Asterisk performs well on Gentoo too, it would
be great to see if any customized installation has been made for
asterisk, to get it in a min. resource/full throttle
Hi folks,
Just wondering if someone have already done something like that :
SIP Client_A ---1)call--- ASTERISK ---2)outgoingcall-PSTN--Client_B
|
|
Use the manager interface:
Action: Originate
Exten: 8320 # meetme room extension in
extensions.conf
Channel: Zap/g1/3125551212 # outside number to dial
Context: default
Priority: 1
MATT---
-Original Message-
From: Areski [mailto:[EMAIL PROTECTED]
I am doing debug on the voip rtp session named-events. The router only does
RTP encoding for the dtmf on calls that originate from an FXS interface on
the router - not from a phone running SCCP that is connected to the router
(Cisco CallManger Express / ITS). However, if I call one of the FXS
*CLI NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration
from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.70'
I've seen this in only two cases:
1) when the SIP user ID, the Authenticate ID (on the GS) and the
extension name in sip.conf are not all the same, in your
Hi,
I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to
work againts Asterisk; at least there is some MGCP conversation between
them but when I offhook a phone attached to IAD I get no tone at all.
As anybody managed to get working Asterisk against an MGCP Cisco
You can't use INFO with the GS. * and GS interpret the INFO standard
differently. As a result, the GS does multiple digit transmission. Use
either inband or the RFC2833 option. INFO will not work no matter what
you do.
I am using the following settings
Software Version:
Now that you mention it, I did observe the PUBLISH message. Can someone
please tell me exactly the change that was made that fixed this?
(file/lines) I can do a diff -r and see a few changes from CVS but I'd
like to be sure. We have a lot of custom changes as well so it's
non-trivial to update
On Fri, 2003-11-21 at 17:57, TC wrote:
I am using the following settings
Software Version:Program--1.0.4.20Bootloader--1.0.0.12
HTML--1.0.0.19
Where is this firmware from? The GS site is still at
Program--1.0.3.81Bootloader--1.0.0.7HTML--1.0.0.18
or is Sipphone ahead of GS?
TC wrote:
You can't use INFO with the GS. * and GS interpret the INFO standard
differently. As a result, the GS does multiple digit transmission. Use
either inband or the RFC2833 option. INFO will not work no matter what
you do.
I am using the following settings
Software Version:
Asterisk Users
In an attempt to help Asterisk move forward, a number of us have decided
to create a book. It would initially be released as an ebook that
could be sent to newbies to help them up the rather steep learning
curve. Ultimately I would like to see it published and sold in
bookstores
Reason.
I have a fax/ans phone with handset, that lets you monitor the caller, so if
you wish, you can pickup the call.
The asterisk is undergoing testing, it will then be online tested at the house
so I can get more familiar in setting components up, e.g. sip phones,
voicemail, transfers
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Areski
Sent: woensdag 19 november 2003 19:27
To: Asterisk-Users Mailing-list
Subject: RE: [Asterisk-Users] hold music =]
http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20redhat
FYI
Hello,
today I tried a DIAX - * - DIAX connection over the internet (768/128
ADSL connection on both sides).
The sound quality was great. However, we had some latency problems, and
also, if both sides where not talking the first words had some problems
getting thru.
Is this expected, is there
when i connect the phone directly i get the caller id
after the first ring and b4 the second one. what does
that suggest?
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Fri, 2003-11-21 at 08:17, C M wrote:
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Fri, 2003-11-21 at 07:14, C M
New to ADSI but my guess would be...
If you want Asterisk to put the call on hold
you could just program the soft keys to send the DTMF
tones that a regular phone would use to put the call
on hold.
If you look at the example that can with Asterisk.
(/etc/asterisk/asterisk.adsi ..I have been
Hi,
I'm having trouble with asterisk: I can't hear both way of a call.
here is my current architecture:
grandstream - siproxd - asterisk - pstn
As I'm just testing for the moment, evrything is on my LAN. I know that
there is no need to have a proxy here. But Later, the asterisk will be
Help: Checkout as of 17:00 UCT
Does anyone know if:
chan_zap.c: In function `zt_train_ec':
chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this function)
is expected at the moment?
Dave Kitchen
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Especially with respect to a Orieley co book, think about a removable
card that lists in short description the applications and the arguments.
This could be the item that gets taped to the side of the monitor of the
new user trying to lay out his dialplan.
On Fri, 2003-11-21 at 11:34, Steven
Dave,
Help: Checkout as of 17:00 UCT
Does anyone know if:
chan_zap.c: In function `zt_train_ec':
chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this function)
is expected at the moment?
That variable I think is defined in zaptel directory. I had the same problem
and had to
On Fri, 2003-11-21 at 12:03, Dave Kitchen wrote:
Help: Checkout as of 17:00 UCT
Does anyone know if:
chan_zap.c: In function `zt_train_ec':
chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this function)
is expected at the moment?
Dave Kitchen
Google lesson for today
Take the
You mean you get CallerID on a CallerID capable phone, right?
Try this: Load a terminal communications app like minicom, bitcom, telix, or something
you're familiar with. Plug the phoneline into a modem that you can see with your
temimal software. Ring the phone. I believe in the US, you will
On Friday 21 November 2003 12:03, Dave Kitchen wrote:
Help: Checkout as of 17:00 UCT
Does anyone know if:
chan_zap.c: In function `zt_train_ec':
chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this
function) is expected at the moment?
You need to update your zaptel CVS first.
As I understand you have PRI connected to asterisk directly.
In my case there is a h323 gateway between them and the h323
driver must recognize the not in service signal and made asterisk
aware of it so that asterisk could relay the conditions/recorded messages
to SIP phones. From my experience so
Good day,
I have a new Grandstream and am having trouble connecting to *
my software version is the same as below...
I can get it to connect, but am getting RTP Read error: Resource
temporarily unavailable errors whenever I dial...
Tom
- Original Message -
From: Dave Cotton [EMAIL
Especially with respect to a Orieley co book, think about a removable
card that lists in short description the applications and the
arguments.
This could be the item that gets taped to the side of the monitor of
the
new user trying to lay out his dialplan.
I have added the pull-out card to the
Hi Steven,
I think this is a great idea and the best way to make
users more familiar with Asterisk and its configuration and usage.
I can and will provide input for all H.323 related sections.
Michael.
Steven Sokol wrote:
Asterisk Users
In an attempt to help Asterisk move forward, a number of
Hi,
I get a number from DeltaThree and I want to setup on my Asterisk to receive
and to make calls thru this phone #.
I need some samples of .conf with please.
Best regards,
Chris HARIGA
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi,
- Original Message -
From: Peer Oliver schmidt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 21, 2003 7:49 PM
Subject: [Asterisk-Users] DIAX, IAX2 and latency
Hello,
today I tried a DIAX - * - DIAX connection over the internet (768/128
ADSL connection on both
Sounds like a great idea! I'll gladly help if requested (I'm a technical
writer).
Comment: I don't see anything on echo cancellation. That's a big enough and
common enough issue that it deserves some discussion.
--Ernest
At 10:46 AM 11/21/2003, you wrote:
Hi Steven,
I think this is a great
On Friday 21 November 2003 06:17 am, Michael T Farnworth wrote:
On Thu, 20 Nov 2003, marrandy wrote:
host=dynamic
defaultip=192.168.254.160
O.K. - that worked, plus I removed the permit and went back to inband (or
grandstream calls it, in-audio.
So I can make and receive calls.
My next
My issue is programing the ADSI softkeys... But what is the syntax for
Hold. Example to create softkeys to transfer I can send the FLASH key.
But I am lost when it comes to the Hold.
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Biggs
On Fri, Sep 24, 2004 at 07:13:39AM -0400, [EMAIL PROTECTED] wrote:
Yes, it is. But why would you want to do that when yo said what you
want it to be at 6.0.
He's got the Skinny version and wants to change to the SIP version.
Maybe you didn't expling what and why you want to do it in
Software Version:Program--1.0.4.20Bootloader--1.0.0.12
HTML--1.0.0.19
Send DTMF: via SIP INFO
DTMF Payload Type: 101
In my experience, anything higher then 1.0.4.17 will NOT work properly
with asterisk. You can get the phone to sip register but it will not stay
that way.
Great idea!
I would also like to see some discussion of ISDN BRI and CAPI
hardware, as well as some discussion of distributed asterisk
what happens when you start deploying a network of many asterisk
boxes, how to do forwarding and switching properly, TDMoE as well as
E164 enum call routing -- on
Absolutely Right! I just added it to the living outline.
Thanks,
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessenger
Sent: Friday, November 21, 2003 1:07 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Outline For Asterisk
On Fri, 2003-11-21 at 20:14, marrandy wrote:
My next observation, is many of the buttons on the phone don't work e.g.
Called - After make calls, I expected that pressing this button would show
something. Nothing is shown.
Pickup and press called the last 10 numbers called are shown
Title: RE: [Asterisk-Users] Outline For Asterisk Book - Please Review Comment
This is a great development. What a good way to develop a book for a great piece of software! I had an ouline slowly developing as my Asterisk implementations broadened, but now I'll jump on board behind yours. I
If anybody is interested, I have an early version of my Call Manager for
Windows application integrated with Asterisk. CMW is an application bar
(like the task-bar) that docks to the top of your desktop window. It
provides the following functions:
1. View Call-Related Information (Caller ID,
On Friday 21 November 2003 03:22 pm, Dave Cotton wrote:
On Fri, 2003-11-21 at 20:14, marrandy wrote:
My next observation, is many of the buttons on the phone don't work e.g.
Called - After make calls, I expected that pressing this button would
show
something. Nothing is shown.
How about this whole discussion getting a mailing list of it's own. I do
want to contribute, and would join a new mailing list for this, but
would like to take it out of the -users list to cut down on volume and
get to answering questions easier.
On Fri, 2003-11-21 at 11:34, Steven Sokol wrote:
I think this is a great addition!!!
Thanks for the app!
-Original Message-
From: Steven Sokol [mailto:[EMAIL PROTECTED]
Sent: Friday, November 21, 2003 3:32 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1
(Alpha)
If anybody is interested, I
An excellent suggestion. Would Digium be willing to host this, or
should we find a different host? Personally I would like to keep
everything on the Digium server, but I can understand if Mark doesn't
want to cover the additional bandwidth. What do you think is best?
Steve
-Original
Randy,
Great suggestions! Jared is working on integrating the changes since
this morning. I think I may drive him nuts, but we will get these
integrated too. If you get a chance, drop by #asterisk-doc.
Thanks,
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Title: Message
I hope Outlook doesn't mess this up too
much--I've to make it as list-friendly as I know how...
Try
Outlook QuoteFix at http://home.in.tum.de/~jain/software/outlook-quotefix/
It
worked great until this last email :)
Hi All,
I have a grandstream sip phone which I have figured out and configured
to make internal calls. How do I now configure asterisk to allow this
phone to make an outside call?
Thanks in advance,
Steve.
___
Asterisk-Users mailing list
[EMAIL
Steven Sokol wrote:
An excellent suggestion. Would Digium be willing to host this, or
should we find a different host? Personally I would like to keep
everything on the Digium server, but I can understand if Mark doesn't
want to cover the additional bandwidth. What do you think is best?
I
Do you have some sort of device that will allow you to make a call to an outside line?
- Original Message -
From: Steve Bradwell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 21, 2003 4:12 PM
Subject: [Asterisk-Users] making outside call with sip phone
Hi All,
I
Again, can you please confirm you are neither running serial console *nor*
graphical console (e.g. framebuffer). If you can call into the office we
can ssh in and take a look at the configuration.
Mark
On Fri, 21 Nov 2003, Scott Stingel wrote:
(Apologies: starting this as a new thread - I'm
I recently moved my Asterisk configuration to a new server and re-built
Asterisk from CVS. Now, I'm experiencing the following issue with SIP:
Executing Dial(Zap/1-1, SIP/100|20) in new stack
NOTICE[-1232077904]: File app_dial.c, Line 518 (dial_exec): Unable to
create channel of type 'SIP'
A little more about Linux commands to help troubleshoot installations. Also using
gnomephone would be nice.
-- Original Message --
From: Steven Sokol [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Fri, 21 Nov 2003 11:34:35 -0600
Asterisk Users
In
Can I have some installation steps on this
it looks very good..
Alex
Unfortunately I haven't had a chance to work up an install. If you have
the core VB runtime file (MSVBVM60.dll) you will only need a couple of
additional files:
Windows Scripting Runtime: scrrun.dll
MS Winsock Control:
[EMAIL PROTECTED] wrote:
I recently moved my Asterisk configuration to a new server and re-built
Asterisk from CVS. Now, I'm experiencing the following issue with SIP:
Executing Dial(Zap/1-1, SIP/100|20) in new stack
NOTICE[-1232077904]: File app_dial.c, Line 518 (dial_exec): Unable to
create
Will,
I have added Clustering/TDMoE and ENUM as two additional topics in the
Advanced Configuration section. I have also added the ISDN BRI/CAPI
stuff to the Add-On/Optional Components/Hardware chapter of Section 1.
Thanks for the suggestions.
I would love to have whatever you can write up on
Correct, never the less I do not see the same issues
other are reporting with SIP not staying registered
I only saw this after I had installed 20 phones. When I had only one
phone installed during the test and setup phase, it never happened.
Either it is rather infrequent, or you have the
I think the script host gets installed with Windows explorer. If you
don't have it, you can use the DLL in the dlls download:
http://www.sokol-associates.com/Downloads/Dlls.zip
Hope that helps.
Thanks,
Steve
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
WipeOut [EMAIL PROTECTED] said:
I would recommend you dump i4l and use a CAPI card with the chan_capi
driver.. The cheap solution is a AVM FritzPCI card(this is what I use)..
The other solution is the either the Eicon or AVM active cards..
I have experienced lots of bus hangups with the
I've got a couple of PRIs coming in from a SUMA 4 switch with some 800
numbers routed through it.
When the calls come in, I get the following message on the console and the
call never makes it through:
(800 number is fake)
Extension '8005551212' in context 'nonauthenticated' from '232102749585'
check 'show dialplan nonauthenticated'
regards
Martin
On Fri, 21 Nov 2003, James Sharp wrote:
I've got a couple of PRIs coming in from a SUMA 4 switch with some 800
numbers routed through it.
When the calls come in, I get the following message on the console and the
call never makes it
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