Hi guys, I've been using a Polycom IP 301 for a couple of weeks now and
find that it's extremely slow for configuring. For instance, it takes
several minutes to boot up, apply any changes via the web interface takes
at least a minute, etc. Is this normal behaviour? Is there anything that
can be
hi,
i've just upgraded to latest trunk. everything compiles fine but when
starting this message appears and fails to start.
WARNING[3990] loader.c: module chan_zap.so error
/usr/lib/asterisk/modules/chan_zap.so: undefined symbol:
ast_pickup_call
thanks,
paradise dove
_
as of now, 8 ports for 8 phones.
there will be soon a 16 ports version (within April)
On 3/25/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
How many phones lines ?
-Message d'origine-De :
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] De la part de Curt ShafferEnvoyé : vendredi 24 m
On 3/13/06, Alejandro Kauffmann <[EMAIL PROTECTED]> wrote:
> RHEL 4 and therefore CentOS 4 had a bug introduced in the latest kernel.
>
> https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568
>
> This bug report has a typo as well. It should read:
>
> #define DEFINE_RWLOCK(x) rwlock_t x = RW
On Sat, 2006-03-25 at 11:52 +0330, Paradise Dove wrote:
> hi,
> i've just upgraded to latest trunk. everything compiles fine but when
> starting this message appears and fails to start.
>
> WARNING[3990] loader.c: module chan_zap.so error
> /usr/lib/asterisk/modules/chan_zap.so: undefined symbol:
I am having trouble getting DTMF mode to be set to inband on incoming
calls.
I have the following set, and for some reason the connection is still
negotiated with rfc2833.
[outbound]
type=friend
secret=XXX
username=XXX
authuser=XXX
host=XXX.XXX.XXX.XXX
context=inbound
qualify=200
ins
Hello there!
I've problem with setting up unicall / mfcR2.
can't find proper notation for channel, trying unicall/1,
unicall/1/1001, unicall/g1, unicall/g1/1000
and still having no luck.
klaudia*CLI> !cp call /var/spool/asterisk/outgoing
-- Attempting call on Unicall/1001 for application
It is not so bad - it's the docs that aren't so clear - try this:
http://www.oinko.net/astrecipes/index.php?q=astrecipes/understanding+queue+logic
and beware of timeouts
l.
In data Fri, 24 Mar 2006 02:00:10 +0100, Douglas Garstang
<[EMAIL PROTECTED]> ha scritto:
Egads. Getting queues
Currently running fc3 (2.6.9-1.667) on a test box with a TDM04b and
A200D installed with svn trunk. Its working well except one of the cards
is sharing interrupts with another device that I'd like to clean up.
I'd like to try moving to io-apic interrupt support for more interrupts.
The MB bios
Title: Message
As of now we are probably looking in the
36 range. We would like to utilize this as a first step to migrating to a VoIP
system.
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of [EMAIL PROTECTED]
Sent: Saturday, March 25, 2006
2:48 AM
To: 'Asterisk Us
On 3/25/06, Daniel Hazelbaker <[EMAIL PROTECTED]> wrote:
> We are looking at installing a VoIP system with Asterisk and are
> currently looking at the line of 3Com phones. Has anybody had
> success with using the following phones? We need to buy a lot and we
> don't want to end up with ph
Title: Message
Curt Shaffer wrote:
As of now we
are probably looking in the
36 range. We would like to utilize this as a first step to migrating to
a VoIP
system.
To save cost. I would buy a couple of used Adtran 750's, they are cheap
and readily avail
smime.p7m
Description: S/MIME encrypted message
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Hi all ,
I am gettign this warning in my asterisk log after installing g723 codec
:WARNING[3675] samples/codec_g723.c: Received a G.723.1 frame that was
4 bytes from RTP
what that mean ?
thanks
Salaque
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Hi There.
I have the following setup :
Asterisk 1.2.4 , freePBX 2.0.1, spandsp-0.0.2pre24
My problem is as follows :
If I set up a very simple extensions.conf. when I dial from a fax
machine, it seems as if no fax is being recognised.
If I answer the call, I can hear the fax machine beeping.
Hi ll,
anyone never used this QuesCom 400 IP/GSM with Asterisk ? Should it works?
Thanks all
Giordano
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I would not recommend the 3Com phones. I know to get most of them to even
work on 3Com systems you need to purchase licenses. For the prices you want
to pay you would definitely be better off going with something else.
The list price for the 3101 is $155
The list price for the 3102 is $240
The
Giordano Grandis a écrit :
Hi ll,
anyone never used this QuesCom 400 IP/GSM with Asterisk ? Should it works?
Yes, it works fine.
Cheers,
Jean-Michel.
--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT
Ok, perfect. And what protocol is it use ? SIP or SCCP ?
Thanks
-Messaggio originale-
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Jean-Michel
Hiver
Inviato: sabato 25 marzo 2006 14.30
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [Asterisk-Users]
Giordano Grandis a écrit :
Ok, perfect. And what protocol is it use ? SIP or SCCP ?
SIP.
Cheers,
Jean-Michel.
--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
__
Title: [Asterisk-Users] Re: Subscription state after reload (New subject)
We have GXP-2000s and they too will automatically
resubscibe in about an hour.
- Original Message -
From:
William
Harrison
To: asterisk-users@lists.digium.com
Sent: Friday, March 24, 2006 5:28
I am copying the Master.csv file to another server and importing to
mysql. I am looking for a simple billing application that will produce a
bill for a give account code for a give period, based on a rate table.
Is this available?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
I
IF all the things are right, at *CLI> "UC show channels" will show you
all the channels you have.
If in your unicall.conf you wrote "group = 1" along with "channels =>"
declaration, that range will belong to group 1 and you can use them
like Unicall/g1.
How many spans are you using? Are they T1
First of all... why don't you either load cdr_ODBC or cdr_MYSQL and
save yourself the stress of loading the .csv file into a database.
Now that that is done.. it's pretty trivial to write something to
interface with your billing software that will look at minutes used
and bill the customer.
That
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Disclaimer: astpp is my software. :-)
It's quite easy to do this with astpp. Depending on exactly what you
want, there are a few ways to do it. Drop a note on the astpp forum
(www.astpp.org/forum) or on the astpp mailling list if you're interested.
Yes.
My phone's context within sip.conf has
[EMAIL PROTECTED]
to tie the sip registration from the phone to the sip peers in *
On Mar 24, 2006, at 7:28 PM, Nicholas Kathmann wrote:
William M Conlon wrote:
sip debug enable (or whatever the CLI commad is)
This will show you the message for y
I'vce got three Asterisk systems here that I'd like to be able to place
calls between with IAX. Each system, pbx1, pbx2, pbx3, should be able to
connect to every other. Do I need separate user/peers or can I combine them
into a single user=friend for each system?
Has anyone got a working example t
2 x TDM2460E (with hardware echocan module) or TDM2460B (wo/echocan) = 48
phones lines, no T1 cards, no channel banks level adjustments troubles,
direct Zap channels and simple switching.
Probably the best choice and price :-)
Best Regards,
Francois BERGERET,
France.
A very happy TDM2400 user
12
hours later... still playing with this. Anyone got any
ideas?
Doug.
-Original Message-From: Douglas Garstang
Sent: Friday, March 24, 2006 10:53 PMTo: Asterisk Users
Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List -
Non-Commercial Discu
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: Saturday, March 25, 2006 12:27 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE : RE : [Asterisk-Users] FXS channel banks
>
> 2 x TDM2460E (with hardware echocan module) or TD
No, you need to use different names. You
can use friend rather than having separate entries for in/out. What do you get
when you type iax2 show peers? You should be able to use friend and the same
three entries on each box with the exception of changing the IP addresses.
Fr
Darren Wiebe napisał(a):
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Disclaimer: astpp is my software. :-)
It's quite easy to do this with astpp. Depending on exactly what you
want, there are a few ways to do it. Drop a note on the astpp forum
(www.astpp.org/forum) or on the astpp mailling
Well,
right now I have this on box1:
[pbx1]type=friendauth=rsainkeys=pbx1outkey=pbx1context=global_pbx_transferhost=pbx1.ipt.yyy.comdeny=0.0.0.0permit=xxx.187.142.203
[pbx2]type=friendauth=rsainkeys=pbx2outkey=pbx1context=global_pbx_transferhost=pbx2.ipt.yyy.comdeny=0.0.0.0permit=xxx.187.14
Darren Wiebe wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Disclaimer: astpp is my software. :-)
It's quite easy to do this with astpp. Depending on exactly what you
want, there are a few ways to do it. Drop a note on the astpp forum
(www.astpp.org/forum) or on the astpp mailling list
Actually, I commented out [pbx3] on the caller, and the callee is STILL
seeing pbx3 as the username. That's even more bizarre. I am sendng pbx1 as the
key from the pbx1 system, and pbx2 is matching it against pbx3???
Huh???
-Original Message-From: Douglas Garstang
Sent: Saturday
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I didn't want to bore everybody with the details but I'll try. :-)
A few questions/comments:
1. Are you wanting invoices to print or email or do you want
something that keeps track of a prepaid balance? We can do both.
ASTPP itself will handle the s
This
is INSANE! My calling system has this iax.conf:
[pbx1]type=friendauth=rsainkeys=pbx1outkey=pbx1context=global_pbx_transferhost=pbx1.ipt.yyy.comdeny=0.0.0.0permit=xxx.187.142.203
[pbx2]type=friendauth=rsainkeys=pbx2outkey=pbx1context=global_pbx_transferhost=pbx2.ipt.yyy.comdeny=0.0.0.0p
Douglas Garstang wrote:
This is INSANE! My calling system has this iax.conf:
Search the archives for mails about separating originations/terminations
by removing all friends and setting up the various interoperating boxes
in a peer-user arrangement.
I am pretty certain there are archived
There are some companies that have a basic GUI. Try emailing the Biz. list. DovidJustin Hamade <[EMAIL PROTECTED]> wrote: You will probably have to build that yourself, or really customizesomething off the shelf. Depending on what phones you are using youmight be able to do that via the phones
Just a few things Doug and they are just
constructive criticism so don’t take them the wrong way.
You
hijacked some else’s thread about a SIP trunk problem. Very
frowned upon and will decrease people willing to help..
All
of your posts are so dramatic and many times
At 06:25 PM 03/23/2006, you wrote:
Perhaps I'll stay w/ Vonage for the time being - I hate having it go
analog and back to digitial in the space of 5 feet of cable, but at
least it works well! (Flawlessly as a matter of fact, in the 14
months I've had it!!)
When I took the 3 feet of analog c
Well, I just tried your approach. I broke them all up into users/peers. Now it
makes even LESS sense. The pbx1 system is connecting to the pbx2 system, and
according to the iax debug, is sending a username of 'pbx3_in'. *lol*
[pbx1_in]
type=user
auth=rsa
inkeys=pbx1
context=global_pbx_transfer
You do realize you're not sending along a username so it's using another method
to try to discover the username you're trying to authenticate as on the server
side? Apparently not.
IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]
Joshua Colp
- Original Message -
From: Douglas Garstang
[mailto:
On Mar 25, 2006, at 2:01 PM, Steve Totaro wrote:
Just a few things Doug and they are just constructive criticism so don’t take them the wrong way. 1 You hijacked some else’s thread about a SIP trunk problem. Very frowned upon and will decrease people willing to help..
This does not appear to be
Why do I need a username at all if I am doing rsa authentication? Why doesn't
it match against the key?
> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Saturday, March 25, 2006 12:11 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE:
It still needs to know the username so it knows what entry in iax.conf to use
for that information, such as the key to use.
Joshua Colp
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECT
Well,
actually I did... sort of... I picked a random post and hit reply as I normally
do. I forgot to clear the old text, but it's obvious that in no way it detracted
from the other person's post, and the use of the word 'hijack' is highly
dubious..
Doug.
-Original Message-From:
I could ask why it can't authenticate against the key, but we've already been
there.
So, if I have 5 asterisk systems, and I want to have a different key on each,
and each system has a user and a peer section, and I have to use different
usernames... oh boy... this sounds like a horrible mess.
Douglas Garstang wrote:
Well, actually I did... sort of... I picked a random post and hit reply
as I normally do. I forgot to clear the old text, but it's obvious that
in no way it detracted from the other person's post, and the use of the
word 'hijack' is highly dubious..
Dubious to you or
Not to hijack but…
http://en.wikipedia.org/wiki/Thread_Hijacking
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
Sent: Saturday, March 25, 2006
2:19 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE:
IAX Incoming/Outgoing
On Sat, 25 Mar 2006, Nick Hoffman wrote:
Hi guys, I've been using a Polycom IP 301 for a couple of weeks now and
find that it's extremely slow for configuring. For instance, it takes
several minutes to boot up, apply any changes via the web interface takes
at least a minute, etc. Is this normal b
As far as I can tell everything is pretty much the same. Below is the
debug output for a particular phone I left a voicemail for. Maybe I am
missing something that I am just not seeing. Otherwise I'm still not
getting a count, but the other notifications are still working.
Thanks again,
Kevin
Has or does anyone use Gafachi’s (www.gafachi.com) origination (and
termination services)? If so, what can you tell me about their call quality,
and have you had any problems with the service?
Comments are appreciated!
Thanks,
Wes Baehr
Ability Business Computing, Ltd.
Offic
On Mar 25, 2006, at 11:18 AM, Douglas Garstang wrote:
Well, actually I did... sort of... I picked a random post and hit reply as I normally do. I forgot to clear the old text, but it's obvious that in no way it detracted from the other person's post, and the use of the word 'hijack' is highly dub
On Fri, 24 Mar 2006, Usman Tahir wrote:
For the conf on Xfer issue, use the latest beta
http://fox.snom.com/download/snom360-5.5.1b-beta-SIP-j.bin
what's the changelog for 5.5.1b?
-Dan
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Aster
Hi Adibar,
Any success with the gsm gateway?
I have exactly the same problem with units received this month.
The codes given by Sam are not working...
Please, let me know if you have discovered something.
Thanks in advance,
Benchev
> But these are the wrong instructions again. Same as those
> ones
Nick Hoffman wrote:
> Hi guys, I've been using a Polycom IP 301 for a couple of weeks now and
> find that it's extremely slow for configuring. For instance, it takes
> several minutes to boot up, apply any changes via the web interface takes
> at least a minute, etc. Is this normal behaviour? Is
In my expirience the Broadvoice has a lot of "audio issues". A number of times DTMF did not work. You pretty much get what you pay for.Will Glass-Husain <[EMAIL PROTECTED]> wrote: Hi,I'm struggling a bit with DTMF. It seems to work fine on my internal network, but when I call outside lines with
Melcon Moraes napisał(a):
IF all the things are right, at *CLI> "UC show channels" will show you
all the channels you have.
If in your unicall.conf you wrote "group = 1" along with "channels =>"
declaration, that range will belong to group 1 and you can use them
like Unicall/g1.
How many span
Hi Guys,Im having a problem compiling zaptel 1.2.4 on CentOS 4.3, anyone encountered this problem before?Here's the error i got:make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules
make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /usr/src/zapt
Why are you using the bit mask 31? I mean, Do you really need ANI after
DNIS and all the other stuff?
So far, thats all about zaptel setup. Just to make sure, did you really
have chan_unicall loaded? What's the output of "UC show channels".
If everything is allright, you should be able to dial li
http://bugs.digium.com/view.php?id=6696
[]'s
MM
-Original Message-
From: "Mark Quitoriano" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
, "CentOS mailing list" <[EMAIL PROTECTED]>
Cc:
Sent: Sun, 26 Mar 2006 05:39:44 +0800
Delivered: Sat
Melcon Moraes napisał(a):
Why are you using the bit mask 31?
No, but i've tried every combination there. No diffrence there.
So far, thats all about zaptel setup. Just to make sure, did you really
have chan_unicall loaded? What's the output of "UC show channels".
*CLI> uc show channe
Looking for a Tech. that could install and configure Asterisk
systems in and out of California per job basis?
Mark
Voice international
714-279-0204 ext 102
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To
When our tech support answers a call from Queue() and puts the caller on
hold, after 5 minutes the Queue() exits and they are dumped back out.
How can I prevent this? I want our tech support people to be able to put
callers on hold as long as needed.
-Dan
_
On Fri, 24 Mar 2006, Brian Kennedy wrote:
Anyone have a Snom they're happy with? How did you manage that? :)
I would be happier if snom fixed the US indications and the giant 3000
point font they use for everything.
-Dan
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Hey William,
Yes, Mercury-Network-Emp is the context of my voicemail.conf, which is
why in the sip it has the @Mercury-Network-Emp so it knows which context
to apply it to. Any other ideas?
Thanks,
Kevin
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Maybe it does mater, while starting asterisk -vvvc:
[chan_unicall.so] => (Unified call processing (UniCall))
== Parsing '/etc/asterisk/unicall.conf': Found
Loading protocol mfcr2
Mar 25 22:17:23 WARNING[13951]: chan_unicall.c:634 unicall_report:
MFC/R2 UniCall/1 Call control(9)
Mar 25 22:17:2
Yes,There are issues witht he latest kernal release. Search the list archives. DovidMark Quitoriano <[EMAIL PROTECTED]> wrote: Hi Guys,Im having a problem compiling zaptel 1.2.4 on CentOS 4.3, anyone encountered this problem before?Here's the error i got:make -C /lib/modules/2.6.9-34.EL/bui
Maybe I could help. Located in Buena Park
Bart
- Original Message -
From:
[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, March 25, 2006 2:06
PM
Subject: [Asterisk-Users] On site
installtion Tech. wanted
Looking for a Tech. that
Hi;
Loock for ur rtpholdtimeout and rtptimeout in sip.conf.
2006/3/25, [EMAIL PROTECTED] <[EMAIL PROTECTED]>:
When our tech support answers a call from Queue() and puts the caller onhold, after 5 minutes the Queue() exits and they are dumped back out.
How can I prevent this? I want our tech
Has anyone ever gotten * to work on commercial unixes such as HP-UX,
Solaris, AIX?
What about other architectures than x86?
--
.
-BEGIN GEEK CODE BLOCK-
Version: 3.1
GCS/GIT> d-@ s+:+ a? C+++ BLHIS$ U+++ P+> L+++ !E W+++$ N++ o+ K w--
PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+> h
On Mar 25, 2006, at 2:59 PM, Derek Whitten wrote:
Has anyone ever gotten * to work on commercial unixes such as HP-UX,
Solaris, AIX?
What about other architectures than x86?
Yes, i am suing OSX on PowerPC. Works great, but no Zaptel support
due to differences in PCI technology (ie open fir
On Mar 25, 2006, at 4:07 PM, Martin Joseph wrote:
On Mar 25, 2006, at 2:59 PM, Derek Whitten wrote:
Has anyone ever gotten * to work on commercial unixes such as HP-UX,
Solaris, AIX?
What about other architectures than x86?
Yes, i am suing OSX on PowerPC. Works great, but no Zaptel suppo
Hi, all
I have a license for G.729A codec from Digium.
When asterisk starts it shows:
Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:460 load_module: G.729
transcoding module Copyright (C) 1999-2005 Digium, Inc.
Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:461 load_module: This
module is supplied under
Hi there,Im getting this notice in CLI, but the call quality is okey, Im using digium TE406 and asterisk 1.2.4.here are the CLI actual logs: -- Executing SetAccount("Local/[EMAIL PROTECTED]
,2", "XX") in new stack -- Executing AGI("Local/[EMAIL PROTECTED],2", "call_log.agi|50015308467418"
Turn on message threading in your email client and you'll see just how
wrong this is.
Douglas Garstang wrote:
Well, actually I did... sort of... I picked a random post and hit reply
as I normally do. I forgot to clear the old text, but it's obvious that
in no way it detracted from the other pe
Hi, all
I stopped receiving messages from the list. tried to change the
address, gettimg confirmation, but no messages!. All addresses I use
are via gmail. I can see my messages reach the list (looked in
archive), but nothing in e-mail (nothing in spam folder either). Was
ok until sometime ago.
I
Type "show g729" in the CLI. It should show something like this:
0/0 encoders/decoders of 1 licensed channels are currently in use
This will show you how many licensed channels you have and how many are in
use. This might give you a starting point to figure out your problem.
- Gabe
- Ori
The call quality was decent but had to cancel because the 800 DID they
assigned me was not working for incoming calls from Verizon, SBC and
Bellsouth. I was not going to wait around for them to fix it. I
switched to Nufone instead.
On 3/25/06, Wes Baehr <[EMAIL PROTECTED]> wrote:
>
>
>
> Has or
Martin Joseph wrote:
>
> On Mar 25, 2006, at 4:07 PM, Martin Joseph wrote:
>
>>
>> On Mar 25, 2006, at 2:59 PM, Derek Whitten wrote:
>>
>>> Has anyone ever gotten * to work on commercial unixes such as HP-UX,
>>> Solaris, AIX?
>>>
>>> What about other architectures than x86?
>>>
>> Yes, i am sui
Martin, i guess im in dumb mode today because i don't get what you say, may also be because this will be my first welltech to configure.what im trying to do is:remote_voip_gateway>asterisk--->fox/fxs/and_international_voip_providers
call will always go -> this way, no incoming
On Sat, 25 Mar 2006, amer karim wrote:
Hi;
Loock for ur rtpholdtimeout and rtptimeout in sip.conf.
Global Signalling Settings:
---
Codecs: none
Relax DTMF: No
Compact SIP headers:No
RTP Timeout:0 (Disabled)
RTP Hold T
Martin Joseph wrote:
On Mar 25, 2006, at 2:59 PM, Derek Whitten wrote:
Has anyone ever gotten * to work on commercial unixes such as HP-UX,
Solaris, AIX?
What about other architectures than x86?
Yes, i am suing OSX on PowerPC. Works great, but no Zaptel support due
to differences in PCI t
Derek Whitten wrote:
Has anyone ever gotten * to work on commercial unixes such as HP-UX,
Solaris, AIX?
What about other architectures than x86?
Asterisk certainly compiles on Solaris x86. I haven't owned sparc
hardware in years, so I can't tell you firsthand if that works or not.
I rememb
On Mar 25, 2006, at 6:26 PM, Erick Perez wrote:
Martin, i guess im in dumb mode today because i don't get what you say, may also be because this will be my first welltech to configure.
what im trying to do is:
Sorry, I don't know if the 3804 actually uses a similar config setup to my 3701a, so i
Not really. This is just the normal R2 stuff. Except that you may want
to set the tones to match your country's standard, by adding
"supertones=pl" inside your unicall.conf
I was looking at your .call file output and I think that there's
something missing, isn't? What are you trying to do?
The .
Kristian Kielhofner wrote:
> Asterisk certainly compiles on Solaris x86. I haven't owned sparc
> hardware in years, so I can't tell you firsthand if that works or not. I
> remember seeing some stuff in the Makefile for sparc, but that doesn't
> mean that it actually works :).
Yes, Asterisk compi
Hi Dan,
Paste some of your queues.conf and extensions.conf regarding to queue
and also some CLI> output.
[]'s
MM
-Original Message-
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Sent: Sat, 25 Mar 2006 19:32:46 -0800 (PST)
Deliver
Hi Dan,
What does the CLI> says when the users get "call cannot be completed as
dialed..." message?
[]'s
MM
-Original Message-
From: Dan Batrams <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Cc:
Sent: Fri, 24 Mar 2006 18:03:46 -0800 (PST)
Delivered: Fri, 24 Mar
On 03/24/06 07:39 Larry Alkoff said the following:
That's how I _thought_ it worked but extens in such a created
[context_name] are not seen or used by Asterisk to dial out.
There is something missing.
have you included the new context in the context where your phones are set to ?
include
Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel) no hardware interfaces installed gives me this error. Im a bit new to this so any help will be appreciated. == Parsing '/etc/asterisk/musiconhold.conf': Found
Mar 26 00:58:49 WARNING[5171]: res_musiconhold.c:833 moh_register: Unable
I'm pretty sure I already know the answer to this, but...
Is there a way to copy/transfer/replicate sip subscriptions from one asterisk
system to another, for the purposes of HA? You coudln't even write a script to
do it I don't think. You can do an 'asterisk -rx sip show subscriptions' but
th
Have you verified that ztdummy is loaded?
On Sun, 2006-03-26 at 01:06 -0500, Erick Perez wrote:
> Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel) no
> hardware interfaces installed gives me this error. Im a bit new to
> this so any help will be appreciated.
>
> == Parsing '/et
queues.conf:
; How long do we let the phone ring before we consider this a timeout...
;
;timeout = 15
;
; How long do we wait before trying all the members again?
;
;retry = 5
[sales]
strategy = ringall
timeout = 300
retry = 10
member => SIP/1030
member => SIP/4000
member => SIP/4010
--- JR Richardson <[EMAIL PROTECTED]> wrote:
> You certainly can have in/out trunks for each and
> work fine plus you can get
> more granular with security.
>
> If all these boxes are on the same subnet and secure
> from a public network,
> there is really no reason to have an IAX [context]
> for
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