Hi All,
I am trying to call my own service through Asterisk and the DTMF is not
recognized . I also noticed the following issue, the phone rings for about
8-9 times before the line is picked up but when it is picked up it seems
that our system has picked up the call much earlier, I could just no
On Tue, Nov 2, 2010 at 8:29 PM, Dan Journo
wrote:
> Or does this kind of thing need a serious network switch?
>
Why not set MLPPP, assuming your provider supports it.
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Paul Belanger | dCAP
Polybeacon | Consultant
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Blog: http:/
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
> Sent: Wednesday, November 03, 2010 7:28 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] doh! chan_dahdi.conf
>
On Wed, Nov 03, 2010 at 06:30:09PM -0400, William Stillwell (Lists) wrote:
>
>
> For those who don't know, (as I just figured out by reading the sourcecode),
> that all settings for a particular "channels" must be placed before the
> channel => entry.
>
> Immediate=no
> Channel=>1-24
> Immediat
For those who don't know, (as I just figured out by reading the sourcecode),
that all settings for a particular "channels" must be placed before the
channel => entry.
Ie,
Immediate=no
Channel=>1-24
Immediate=yes
Channel=>25-48
Immediate=no
Channel=>49-72
1-24 will have
For those of you who may have missed the announcements made last week at
AstriCon 2010, the Asterisk and Asterisk SCF projects now have a Wiki
site available at
https://wiki.asterisk.org
This site contains a great deal of Asterisk documentation, development
plans and other content, with more to c
You can't do "allow=" then "disallow=all". This will disable all the
codec. Try:
disallow=all
then
allow=g729
allow=ulaw
On 10-10-29 03:37 AM, Mert Hakk? Bingöl wrote:
Hi,
No matter I try, I can not register to Voipwise with Trixbox. It is
always in "unregistered" state in sip registry. He
On Wed, Nov 3, 2010 at 4:32 PM, Danny Nicholas wrote:
> TAM, Bob! Guess I've got to go through now and "unquote" my literals...
Hi Danny,
Glad that helped. But on second thought, maybe the better fix is to
remove the double quotes in the Gotoif()'s, like this:
exten => s,n,Gotoif($[${TEST_RET
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Beers
Sent: Wednesday, November 03, 2010 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Gotoif changed in 1.8
On Wed, Nov 3, 2010 at 4:05 PM, Danny Nicholas wrote:
> Hi Gang,
>
> I’m testing 1.8.0 on one of my machines and this snippet
> “chokes” on line 7 (works fine with 1.4.30)
>
> [tb-account-balance]
>
> exten => s,1,Set(BALCOUNT=0)
>
> exten => s,n,NoOp(Verbose(acct ${digitacc} pwd ${di
> I use the SG560 (http://www.snapgear.com/index.cfm?skey=1557) to do this.
Thats perfect. Any idea where they are available? I cant locate a store online.
Thanks
Dan
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Hi Gang,
I'm testing 1.8.0 on one of my machines and this snippet
"chokes" on line 7 (works fine with 1.4.30)
[tb-account-balance]
exten => s,1,Set(BALCOUNT=0)
exten => s,n,NoOp(Verbose(acct ${digitacc} pwd ${digitpwd} ))
exten => s,n(runagi),Set(TEST_RETURN="NONE")
exten =>
s,n,
El 03/11/10 10:44, Bryant Zimmerman escribió:
I have used 1.4 & 1.6. I am testing 1.8 for production and it is
looking very good. I am making some changes to accommodate some minor
dialplan changes from 1.6. Our 1.4 is very solid 1.6 has some issues
with DTMF issues when used with Sonus on the
Thanks a lots Bryant,
I would test 1.8 and see if it work out, Definitely 1.8 going to be rock sooner
or later, Let's try 1.8
Currently we are facing some issue with echo in conference call with 1.2
version hopefully it will go away with 1.8
Thanks,
S. Patel
From: brya...@zktech.com
To:
Its good to know the MATH function because it can do much more and also deal
with floating point numbers. However in your case a simple addition would be
suffice as other posters posted, or try Danny's GotoIf if it fits your
scenario.
Set(vgLabel=vg${MATH(${vg}+1,i)})
Zeeshan A Zakaria
--
www.il
If 1.2 is working fine without any problem then why do you need to upgrade
to any newer version? I would suggest don't do it. If you really want to do
it just for the sake of doing it, upgrade to 1.4 only, which is the most
stable and well tested version of asterisk. Upgrading always causes hickups
I have used 1.4 & 1.6. I am testing 1.8 for production and it is looking
very good. I am making some changes to accommodate some minor dialplan
changes from 1.6. Our 1.4 is very solid 1.6 has some issues with DTMF
issues when used with Sonus on the back end. 1.8 is looking very good and
we hope
On Wed, 3 Nov 2010, Philipp von Klitzing wrote:
> Hi!
>
>>> Side note: Stay away from solutions that use mISDN, instead go with
>>> Zaptel (DAHDI), Woomera or CAPI.
>>
>> Interesting.
>>
>> I've been usng mISDN for some years now without issues. Why should I
>> migrate to DAHDI?
>
> None - if you
Well the problem seems to be:
the linphones are listening on port 5062, while * is on port 5060. For
some reason, the INVITEs are received from *, but are forwarded on port
5060 by default.
I "solved" the problem by moving * to port 5062 and moving the linphones
back to port 5060. All is well,
Hi!
> > Side note: Stay away from solutions that use mISDN, instead go with
> > Zaptel (DAHDI), Woomera or CAPI.
>
> Interesting.
>
> I've been usng mISDN for some years now without issues. Why should I
> migrate to DAHDI?
None - if you are happy then don't touch it. :-) Otherwise search this
Thanks for reply,
I believe we have around 300 SIP phone register on asterisk and we have 2 T1
line. Roughly i would say max concurrent number 20/30 Max.
My only concern is stability after whatever version migration. I believe 1.8
is new and it's just coming out form egg so quite worry abou
9] NoOp("SIP/123-00075448", "Using
CallerID "123" <123>") in new stack
-- Executing [8613430491...@from-internal:2] Set("SIP/123-00075448",
"_NODEST=") in new stack
-- Executing [8613430491...@from-internal:3] Macro("SIP/123-000754
Am 03.11.10 15:14, schrieb satish patel:
>
> Hello Everyone,
>
> We are running asterisk 1.2.x version in production environment since last 5
> year and we have no issue at all, But now time to upgrade. and i heard about
> 1.8 which has introduce many features. I am wondering should I use aster
Hello everyone!
I've had this problem for a while and cant figure it out. When an outside
caller calls an extension on my asterisk system, they do not hear any sort
of ringing. Inside extensions calling other extensions do hear ringing. We
have 3 other asterisk systems that are configured the same
On Wed, 3 Nov 2010, Philipp von Klitzing wrote:
> Side note: Stay away from solutions that use mISDN, instead go with
> Zaptel (DAHDI), Woomera or CAPI.
Interesting.
I've been usng mISDN for some years now without issues. Why should I
migrate to DAHDI?
Gordon
--
_
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Wednesday, November 03, 2010 9:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Migration from 1.2 to 1.8 in production
Hello Everyone,
W
How many lines are we talking here?
Get a two port T1/PRI Card, use a channel bank, and get your lines from your
provider on a PRI. (this way you can start off with 10 numbers, and add up
to 300+ and never have to add any "extra" lines at a per line price.
If you looking to save money with SIP pr
Philipp von Klitzing said at 03/11/2010 14:10:
> Hi!
Hi :-).
>>> 2. Add BRI card(s) to the computer to run Asterisk and somehow hook
>>> up the Samsung.
>>
>> Do-able. Connect Asterisk to your ISDN2, then host the Samsung off the
>> asterisk box. But then, might as well dump the Samsung and just
Roger Burton West said at 03/11/2010 12:48:
> On Wed, Nov 03, 2010 at 12:05:51PM +, Ronny Adsetts wrote:
>> What hardware would I need in the Asterisk so I could hook up some
>> analogue extensions? Am I right in thinking I need something like
>> an FXO/FXS card?
>
> Yes, this ought to work. I
Hello Everyone,
We are running asterisk 1.2.x version in production environment since last 5
year and we have no issue at all, But now time to upgrade. and i heard about
1.8 which has introduce many features. I am wondering should I use asterisk 1.8
in production ? or should I go with 1.4 or 1
Hi!
> > 2. Add BRI card(s) to the computer to run Asterisk and somehow hook
> > up the Samsung.
>
> Do-able. Connect Asterisk to your ISDN2, then host the Samsung off the
> asterisk box. But then, might as well dump the Samsung and just put
> VoIP phones on everyones desks.
If you decide to go
Hi!
> It is causing an issue for me. One SIP UA works fine - ring, forward, etc.
> While the other does not.
Make the UAs listen on different ports (for example 5060 and 5062) and
see if that solves your problem - if you can't make them have different
IPs, that is.
Also be sure to fully under
> exten => s,n,Set(vgLabel=vg(${number}+1))
> exten => s,n,GoTo(${vgLabel})
>
> But in stead of vgLabel becoming the SUM of 2 numbers, it is just a
> string :
Use the MATH function.
Philipp
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-- Bandwidth and Colocation P
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, November 03, 2010 8:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to make the sum of a ${VARIABLE}
On Wed, Nov 3, 2010 at 9:18 AM, Jonas Kellens wrote:
> exten => s,n,Set(vgLabel=vg(${number}+1))
>
exten => s,n,Set(vgLabel=vg$[${number} + 1])
untested
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) |
Blog: http://blog.polybeac
Hello,
I have this in my dialplan :
exten => s,n,Set(vgLabel=vg(${number}+1))
exten => s,n,GoTo(${vgLabel})
But in stead of vgLabel becoming the SUM of 2 numbers, it is just a string :
[Nov 3 16:17:27] -- Executing [...@macro-f:43]
Set("SIP/test-0002", "vgLabel=vg(1+1)") in new stack
insecure=very should fix it.
On Wed, Nov 3, 2010 at 4:08 AM, Gregory Malsack wrote:
> Can anyone tell me why my inbound calls keep getting rejected with 401?
>
>
>
> Here’s the debug information:
>
>
>
>
>
> <--- SIP read from UDP:147.135.32.221:5060 --->
>
> INVITE sip:6087294...@216.26.109.22:5
On Wed, Nov 03, 2010 at 12:05:51PM +, Ronny Adsetts wrote:
>What hardware would I need in the Asterisk so I could hook up some analogue
>extensions? Am I right in thinking I need something like an FXO/FXS card?
Yes, this ought to work. If you're plugging phones into the Samsung it's
providi
>
> I have sent an e-mail to this list (awaiting moderator approval by the
> size) talking about some difficult to make calls with a SIP Provider in
> Brazil.
> I'm new at this list and have no sure if I have posted my question in the
> right place.
> If this is not the channel to make this kind o
Thanks everyone for your replies so far. I've pretty much concluded that going
for a full Asterisk solution is the best longer term solution and that's what
I'll do. We're moving office before May so that's the perfect time to put in a
new phone system.
But, I need to implement something quick-
On 11/03/2010 03:49 AM, Gordon Henderson wrote:
>>
>> I've got a client with two ADSL connections for redundancy.
>>
>> Is it possible to set up asterisk to connect to one SIP provider using
>> both adsl connections and load balance between the two connections? Or
>> to use one connection as the ma
hi all, please help... I am calling in the simplest way among two
linphone clients connected to one asterisk server... the call ends on
one side without any sign of problem, while on the other side it stays
connected.
I checked the SIP dialogue and at some point the server sends a BYE
message t
Can anyone tell me why my inbound calls keep getting rejected with 401?
Here's the debug information:
<--- SIP read from UDP:147.135.32.221:5060 --->
INVITE sip:6087294...@216.26.109.22:5060 SIP/2.0
Call-ID: 31007e...@147.135.32.221
CSeq: 1 INVITE
From: "Wi M";tag=9bbc
To: "Gregory Malsack"
Via:
On Tue, 2 Nov 2010, Dan Journo wrote:
> Hi,
>
> I've got a client with two ADSL connections for redundancy.
>
> Is it possible to set up asterisk to connect to one SIP provider using
> both adsl connections and load balance between the two connections? Or
> to use one connection as the main one,
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