On Tue, June 27, 2006 0:26, shadowym said:
They have been talking about this for awhile. If you look at the real
time
and embedded operating system world they have not really done so well over
the many years they have been trying. Just throwing money at the problem
has
never worked for them
How? Can u show me?
On 6/27/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Jun 27, 2006 at 12:13:31PM +0800, unplug wrote:
Hi,
How can I access the variable in marco? Say, there is a dial plan
below. In line 4, it will show the variable FOO=1234. However, the
variable in line 2 is
I got a request for one customers to set-up 100 accounts.
I use usually the Caller-ID as the card number.
Is there a way to make it for 100 accounts easier?
To generate 100 cards is not a problem, but if it would work with one
account number would be even better
I could use a different
You have all our respect.
At least mine.
Carry on!
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My teeth are on edge after this one. A couple of perfectly good hours
of my life, and I still don't know what's going on. . . .
The extensions.conf.sample that comes with the current SVN trunk has
this line, in an example that shows how to use ChanIsAvail:
exten = s,n,GoToIf([${AVAILSTATUS}
Brian Capouch wrote:
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
I couldn't get this to work unless I surrounded the first part of the
test with quotes, too, like this:
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
Ooops.
Actually, I mis-pasted one of my
Hello
As far as ive understood, you can just write
Exten = s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail)
${AVAILSTATUS} would return 1, and ${AVAILSTATUS} would return 1
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Brian Capouch
Sendt:
Dear
I am using [EMAIL PROTECTED] , and I have 2 hard disks on the
system ,how can I put the database (CDR) on the second hard disk .
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of
Jon Schøpzinsky wrote:
Hello
As far as ive understood, you can just write
Exten = s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail)
${AVAILSTATUS} would return 1, and ${AVAILSTATUS} would return 1
Through more testing, the double quotes I used seemed superfluous; if
you use them in both
Hello,
We are looking for DID in United
Arab Emirates, Iran,
Kuwaiti, Iraq, Bahrain, Jordan,
Saudi Arabia.
Thanks
Laurent
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Hi again,
the TR6T parameter (i have german settings for my AMO
so it is TR6Q ;-)) resolved the same issue for my...
the difference is that i have an IP-trunk (using
oh323) between Asterisk and the HiPath. Have you tried
to remove the TR6T parameters...
Can you also paste the following outputs
Hello all,I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how?
thanks.Lito
I'm new to this and don't know how to do a sip trace, but have attached the
files as requested.
Thanks for your help.
Dean.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke
Sent: 26 June 2006 15:21
To: Asterisk Users Mailing List - Non-Commercial
Hi again...
normally the 0/16 is a d-channel.
check the config in the zapata.conf. You should have
some thing like this
/etc/zapata.conf
bchan=1-15
dchan=16
bchan=17-31
/etc/asterisk/zapata.conf
channel = 1-15,17-31
i don't rember exactelly but in /proc/zaptel there is
the possibility to
I would guess the card is actually a http://www.x100p.com/products_1.htm
and may be x100p selling it on ebay themselves.
I have one of these cards itself and it works fine. There is a bit of
echo initially but it gets cancelled our fairly quickly. Apart from
turning on echo cancellation I have
Hi all,
Has anybody got an idea if
http://www.globe7.com
supports SIP protocol?
Please send the asterisk config u have it.
Thanks in advance..
Dan
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Hi,
Well to my knowledge, Origination is not legal in these parts of the
world as of now.
Bahrain is open for termination.
http://www.menatelecom.com/products/termination.html
Thanks
Dan
On 27/06/06, Laurent Schweizer [EMAIL PROTECTED] wrote:
Hello,
We are looking for DID in United
Ronald Wiplinger wrote:
I got a request for one customers to set-up 100 accounts.
I use usually the Caller-ID as the card number.
Is there a way to make it for 100 accounts easier?
To generate 100 cards is not a problem, but if it would work with one
account number would be even better
Title: Avaya 4610sw SIP setup problem
Hi all,
I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem!
I have two Avaya 4610sw phones. I installed the latest SIP firmware using
On 26/06/06, Boris Bakchiev [EMAIL PROTECTED] wrote:
Can the TE406P card's VPM module be swapped for the new revision withOctasic chipset?The VPM450M requires a firmware upgrade to the existing base TE2/4XXP cards. This new firmware is known as 3rd Generation firmware. Digium have an upgrade
Title: Re: siemens pbx and asterisk
Hi Lito,
We have successfully integrated an existing Siemens HiPath 4500 PBX with two Asterisk servers.
On the first one we use a H.323 trunk (it needs a card on the PBX, I think it's called HG3550). It works pretty well, except for one missing feature
On 25/06/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- C F [EMAIL PROTECTED] wrote:Neither. It's a separate device, entirely unrelated to any TDM cards (which means it can be used for any type of channel, not just TDM).
The final specs for the number of channels are not yet determined, but
Title: Re: Asterisk x Siemens HiPath 4000
Hi,
Could you post your /etc/zaptel.conf and zapata.conf?
Also, is everything OK the other way round (i.e., from the SIP phones to the PBX)?
Silviu
Hello all.
I have installed and functioning asterisk-1.2.9.1 where I effected one
Hi,
which Hicom and which version is installed?
Hicom 300 or Hicom100?
rich
--- Lito Lampitoc [EMAIL PROTECTED] wrote:
Hello all,
I'm new to asterisk. Our company wants to setup an
asterisk server and will
eventually move to IP centric phones, but they don't
want to just throw away
I am going to try to figure out why mu asterisk box
connected by back to back cable to an PRI appliance is not going to send the
PROGRESSING dss1 message.
In fact i see the SETUP and the follwing CALL
PROCEEDING but not the PROGRESSING so the appliance doesn't allow the "early
audio" !
Hi everybody,
I try this :
[incoming_from_fxo_card]
exten = s,1,Answer()
exten = s,2,Background(filename)
exten = s,3,Dial($(INTERNAL_SIP_TEL))
But * wait the file is finish before make Dial to SIP channel.
Background(filename) (from voip-info.org)
= Starts playing a given sound file, but
I am getting thousand of these messages in asterisk console
Jun 27 12:35:55 WARNING[16496]: codec_gsm.c:194 gsmtolin_framein:
Invalid GSM data
And after some time the system crashes. Does anyone know why?
I running Asterisk SVN-trunk-r7522 built
Does it help to upgrade the system?
Regards,
Hi,
I have the same problem with the queue configuration
When I receive 2 calls only 1 phone ring even if more agent's phone are free.
The second call will go to an other agent only if the first call is pickup.
Somebody have a solution ?This is my config file :Queue.conf[general]
;
Tzafrir Cohen wrote:
On Sun, Jun 25, 2006 at 08:28:35PM +0100, Thomas Kenyon wrote:
I have a TDM400 card with 3x FXO and 1x FXS ports on it.
At the moment I'd prefer (till I can get it working more reliable with
iaxmodem), for a faxmodem to answer one of the lines instead of the
Hi,
Cullin J. Wible wrote:
We have also deployed a dozen of the Linksys SPA-1001 single-line FXS
adapters using G726, SIP, NAT and STUN. They are extremely reliable and easy
to deploy - $60-$70 US each.
I bought a Grandstream GXP-2000 and played now a little with it. It
seems to work really
Hi GL
Pls. config MOH and use Dial command with m option.
This will allow you execute Dial command while providing Music in the
background.
Hope it help
Hoa Thai Duy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, June
Today i put 10 users in a Meetme on a 700MHz machine.
but the result did not satisfy me.
I had all 10 Phones in front of me, cause i'm testing my asterisk.
so i could speak on one phone and listen on any other.
i had a delay of 1 sec of my spoken word(s)
so i think, that you should use a BIG
Thanks for your reply,
But I want to have an interactive menu, not just a music.
So, the customer can have information menu while he's
waiting the call is answer.
I'dont now if it's possible with MoH.
Thanks a lot
-- Initial Header ---
From : [EMAIL PROTECTED]
To
700MHz is very underpowered for a server that will do a lot of meetme.
I would recommend at least a 1.6GHz P4
As for the delay, that problem is usually made worse by using ztdummy
with poor zttest scores. try a different ztdummy timer source or a
hardware zaptel timer if possible because that's
Hello Lito
My PBX HICOM 350 interconnect with asterisk via
Tormenta cards use e1withEDSS1
protocol this work fine
- Original Message -
From:
Lito
Lampitoc
To: asterisk-users@lists.digium.com
Sent: Tuesday, June 27, 2006 11:18
AM
Subject: [Asterisk-Users]
The Wall Street Journal had a write up on this and after reading through it I did not see much in the way of improvement. It seems like the main focus of Micro$oft is to integrate their products with phone systems which can already be done. The article talked about dialing from Office apps and
Silviu, thank's will be this attention. Below my configurations of zapata.conf and zaptel.conf
#zapte.conf
span=1,0,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31loadzone=usdefaultzone=us
#zapata.conf
[trunkgroups]
Hello,
In my asterisk box, i have a zaptel card connected to my analogic pstn line.
I'm using a IAX2 client to call outside :
IAX2 client -- Asterisk -- Zaptel card France telecom line
When checking cdr logs file, i always have an ANSWER on call
status when call on this trunk, even if the
Hi,someone know a good webphone, possibily a free oneThx
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Hi Silviu,
did you manage to get the callername to work? I have a comparable setup
with a hipath System but I can´t get the callername to be displayed over
the trunk. The callernumber works but not the name...
Any suggestion?
Thanks
Michael
We have successfully integrated an existing Siemens
Hi!
I have this setup:
PABX --ISDN30-- Asterisk 1 --SIP-- Asterisk 2 --ISDN30-- TELCO
Digium TE410P is used in both Asterisk 1 and 2.
When I set the CLIR bit on the PABX the Callerid / ANI is removed somewhere between the SIP interface on Asterisk 1 and the SIP interface on Asterisk 2.
I need
Mouss Greg wrote:
Hello,
In my asterisk box, i have a zaptel card connected to my analogic pstn
line.
I'm using a IAX2 client to call outside :
IAX2 client -- Asterisk -- Zaptel card France telecom line
When checking cdr logs file, i always have an ANSWER on call status
when call on
On Tue, 2006-06-27 at 15:00 +0200, Morten Isaksen wrote:
Hi!
I have this setup:
PABX --ISDN30-- Asterisk 1 --SIP-- Asterisk 2 --ISDN30-- TELCO
Digium TE410P is used in both Asterisk 1 and 2.
When I set the CLIR bit on the PABX the Callerid / ANI is removed
somewhere between the
Thanks John for your quick answer,
You're right, i'm trying to put a strong billing system in place, after
many months using ASTCC, i'm now integrating a2billing which seems to
be stronger (my own opinion). But i did'nt clearly undertstand what you
said (my poor english level again ...) Could
On 6/26/06, Peter J Dean [EMAIL PROTECTED] wrote:
I have a issue trying to understand why Asterisk-PBX, when a SNOM
(320 or 360) successfully redirects/diverts a call when it is a local
extension, but fails when you enter external number.
Both the local extension dial and external extension dial
Hi,
As I wrote, the HiPath needs to be upgraded to version 3 (don't ask me any
details, I'm not a Siemens expert) in order to have the CallerID name passed
over the H.323 link. Earlier versions (my case) ony sends and accepts the
CallerId number.
I have set up a workaround for calls coming to
I am in the Cadillac area and would be interested in joining this group
depending on where in the SE it is located and the time the meetings occur.
Thank you,
Jyran Glucky
Advisory Programmer
BlueWare, Inc.
Strategic HealthWare Solutions
3060 W. 13th Street
Cadillac, MI 49601
Phone: (231)
We should figure out what we are doing before we do any investing in domain
names, etc.
I believe that we have the option to be a sub-group off of the glima network if
desired.
http://www.glima.org/autoalley/GLIMA+Network/Member+Benefits/
They are a non-profit org for technology people in
On Tue, 2006-06-27 at 14:03 +0200, Kai Ober wrote:
Today i put 10 users in a Meetme on a 700MHz machine.
but the result did not satisfy me.
I had all 10 Phones in front of me, cause i'm testing my asterisk.
so i could speak on one phone and listen on any other.
i had a delay of 1 sec of my
Hello Silviu,Thank you very much for your reply. I will try that.On 6/27/06, Herchi Silviu [EMAIL PROTECTED]
wrote:
Hi Lito,
We have successfully integrated an existing Siemens HiPath 4500 PBX with two Asterisk servers.
On the first one we use a H.323 trunk (it needs a card on the
How can I limit extension voicemail messages to 10 messages
per user ?
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by
How can I limit extension voicemail messages to 10
messages per user ?
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation
Is it possible to trunk hunt mobile phones in asterisk? say I have one trunkline and 10 mobile phones brought by the engineers in the field, when someone calls the trunkline, asterisk will hunt which of the 10 mobile phones is available. What do I need for this setup?
Thanks in advance.Lito
I think this could be implemented via
follow-me feature of Asterisk
Roger Workman
Business Development
Upperclassman/Universal Holdings LLC
Voice:304.324.3800
Fax:304.324.3801
ICQ: 4447584
Website: http://www.upperclassman.net
Billing Questions: billing at upperclassman.net
HiI have a problem with Dial application. The dialplan looks like this:;exten = x,1,Dial(Sip/|30|L(6:3:1))exten = x,2,Hangup()exten = h,1,DadAGI()
;The call is limited to 60 sec and after that time the conversation stops, but Asterisk never reach the h extension.I
A GSM
gateway will allow you to specify a ruleset so a channel on the gateway is
always locked to a particular mobile number, then you just send the call from
Asterisk to the gateway and it will do the hunt for you.
-Original Message-From: Lito Lampitoc
[mailto:[EMAIL
hi all,
The HG3550 V1 and HG3550v1.1 only supports H.323 V.2.
I'am not sure but i thing that the feature CallerID
Name was introduced in version 3 of the H.323
standard. More informations about the owerviews at
http://www.packetizer.com/voip/h323/.
-Concerning HiPathv3.0.
In version 3.0 the
Andrew Nowrot wrote:
Hi
I have a problem with Dial application. The dialplan looks like this:
;
exten = x,1,Dial(Sip/|30|L(6:3:1))
exten = x,2,Hangup()
exten = h,1,DadAGI()
;
The call is limited to 60 sec and after that time the conversation stops,
but
Khaled Chehab wrote:
How can I limit extension voicemail messages to 10 messages per user ?
If you look in the voicemail.conf.sample file in the source, you can find the
following lines:
; Maximum number of messages per folder. If not specified, a default value
; (100) is used.
Put simply:
Send a call to the PSTN ( analog) through a TDM 400, Sangoma A200, or
an X100 card, once the dialing string is sent, the call will report as
answered.
That's it. You will have no way to PROPERLY bill for the call.
You can assume a call duration of less than one minute ( or
is it possible to route an ISDN-Data channel over an iax-connection ?
the setup is
pc with isdn-card - (zaphfc) Asterisk Server1 (iax) - (iax) Asterisk
Server2 (E1)-connecting to an external isdn-dialin router
via the iax-line the call is transfered as speech which is not accepted at
Hello,
The main differences I can see:
- in zaptel.conf
you have span=1,0,0,ccs,hdb3, which means you ask Asterisk
to serve as a timer for the PBX - on my setup the PBX is the master clock and
Asterisk is the secondary one, so I have span=1,1,0,ccs,hdb3 (in fact, as I use
CRC4 error
Hallo.
I managed to configure asterisk to act as H.323 gateway using asterisk built in
support for H.323. I found it in ./channels/h323 directory of asterisk sources.
I wonder whether asterisk can play a role of H.323 gatekeeper. If Yes, could
You tell me some hints on how to do that.
Hi,
I have an F3000 phone that I am trying to register to asterisk. As
far as I can tell I have everything in correct. Are there any little
quirks I need to worry about? The phone has internet access, set it's
time.. I can access the web config, but it just won't register with
asterisk. I
Greetings all,
Not specifically an asterisk query, but a couple of transfer queries that
I'm sure are obvious to folks who use these phones all the time:
1) how does one do a blind transfer? When a call is answered and one hits
the transfer button, followed by an extension, one has to wait for
On 27 Jun 2006, at 13:54, Il Neofita wrote:
Hi,
someone know a good webphone, possibily a free one
Thx
Ours isn't free - but take a look at
www.mexuar.com , or drop me an email.
Tim.
Tim Panton
[EMAIL PROTECTED]
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[EMAIL PROTECTED] wrote:
is it possible to route an ISDN-Data channel over an iax-connection ?
the setup is
pc with isdn-card - (zaphfc) Asterisk Server1 (iax) - (iax) Asterisk
Server2 (E1)-connecting to an external isdn-dialin router
via the iax-line the call is transfered as
what brand of gsm gateway do you think works well with asterisk?On 6/27/06, Colin Anderson [EMAIL PROTECTED]
wrote:
A GSM
gateway will allow you to specify a ruleset so a channel on the gateway is
always locked to a particular mobile number, then you just send the call from
Asterisk to
Jyran,
We didn't make any plans. Nobody still has confirm anything. All that I did
was a list of everyone that is interested.
I don't know where are we going to meet. If we 're going to meet. Or we may
be do a virtual meeting?
I only need to know if you are interested or not on participate.
What does the CLI show when you make the call? That might help in diagnosingyour problem.
FlynnHi Flynn The situation looks like this:exten = _0800X.,1,AGI(/usr/share/asterisk/agi-bin/checklimit.php|${CALLERIDNUM}|${CONTEXT})exten = _0800X.,2,GotoIf($[${code} = 0
Thanks Steven, This is my first answer.
I'll going to make [EMAIL PROTECTED] the mailing list. I'm going to
include to everyone of the people that has exchange e-mail about this.
After that, we can take care of what to do in advance.
Carlos Alperin
-Original Message-
From: [EMAIL
btw, i got it, 2N Easygate is highly compatible with Asterisk. Thanks.On 6/27/06, Lito Lampitoc [EMAIL PROTECTED]
wrote:what brand of gsm gateway do you think works well with asterisk?
On 6/27/06, Colin Anderson
[EMAIL PROTECTED]
wrote:
A GSM
gateway will allow you to specify a ruleset
Why not add the g parameter and make your deadAGI as the next priority?
I think that would accomplish what you are trying to do.
Example:
exten = x,1,Dial(Sip/|30|gL(6:3:1))exten = x,2,DeadAGI()bp
On 6/27/06, El Flynn [EMAIL PROTECTED] wrote:
Andrew Nowrot wrote: Hi I
I have an Perl AGI script that acts as an IVR for my Asterisk box.
Basically, it simply plays audio files to the caller, collecting DTMF
input and logging every DTMF input into a database table, simply to
document every step or option selected by the caller.
One thing is that in addition
I use
an Ateus VoiceBlue which allows you to do this (never tried it though) which is
a SIP device and you write your dialplan to send calls to the SIP device just
like ringing an extension in Asterisk. It works fine but it tends to drop calls
under load so I have an AGI that determines the
Which public STUN servers are you using or did you setup your own?
-Original Message-
From: Cullin J. Wible [mailto:[EMAIL PROTECTED]
Sent: Monday, June 26, 2006 8:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Discussion'; 'Iain Barker'
Subject: RE: [Asterisk-Users]
On 11:18, Tue 27 Jun 06, William Piper wrote:
Why not add the g parameter and make your deadAGI as the next priority?
I think that would accomplish what you are trying to do.
Example:
exten = x,1,Dial(Sip/|30|gL(6:3:1))
exten = x,2,DeadAGI()
Dont use DeadAGI on
Hello,
Anyone here
have experience with Audiocodes MediaPack MP-108 Gateways?
I would
be willing to pay someone for advice and support with configuring my gateways
for a telemarketing project I am starting. My experience is somewhat limited
but all I want to do is make outbound calls
Is there a way to edit the options available in the voicemail menu trees? My
users are complaining that it's too complicated (I know, it's not really
complicated), and I wanted to remove some of the options if this is
possible. So far I havent' found any info on the wiki or searches, not that
it
Blind transfer is not possible via softkeys on the 7960 using chan_sccp. However, check your features.conf. You should have a line in there regarding blind transfer. I believe the default is #, but it recommended changing that to ##. I did this, and on my 7960s, you hit ## then the extension you
Chan_sccp does not support blind transfer. I would suggest using
chan_sip and the SIP images with these phones; it is much more stable,
has more features and is being actively developed. Chan_sip supports
blind transfer and 3-way calling, plus it handles multiple calls on hold
a bit more
We did it by comment out a number of lines in the code and then re-compiled
just that module.
We also did the same for the company directory.
Other then that I'm not sure if there's much you can do.
Cullin J. Wible
Co-Founder CTO
Email Data Source, Inc.
212-514-8900 x1006
-Original
Thanks for all repliesI noticed that L option does not hangup the call it only limits the call. (In my case the h extension isn't executed). S option can do that (Asterisk reach the h extension)L(x:y:z) - do not hang up the call after x sec.
S(x) - hangup the call after x sec.I also noticed that
Although I've never tried it along withthe L option, you couldtry absolutetimeout:
exten = x,1,AbsoluteTimeout(6)
exten = x,2,Dial(Sip/|30|L(6:3:1))bp
On 6/27/06, Andrew Nowrot [EMAIL PROTECTED] wrote:
Thanks for all repliesI noticed that L option does not hangup the
I'm noticing that the documentation on the voip wiki for voicemail and realtime
voicemail hasn't kept up with reality.
I just created a column called maxmsg in my table. I set it to 1 for the user.
I can leave more than once voicemail message.
Why?
Doug.
We did it by comment out a number of lines in the code and then re-compiled
just that module.
Thx Cullin for the reply, has anyone made a flow chart or end user
instructions for comedian mail? Jus trying not to reinvent the wheel if it's
already been done.
Thanks!
Dan
On 6/27/06, William Piper [EMAIL PROTECTED] wrote:
Although I've never tried it along withthe L option, you couldtry absolutetimeout:
exten = x,1,AbsoluteTimeout(6)
exten = x,2,Dial(Sip/|30|L(6:3:1))I didn't help still the same :(.
-Original Message-
From: Douglas Garstang
Sent: Tuesday, June 27, 2006 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Realtime Voicemail
I'm noticing that the documentation on the voip wiki for
voicemail and realtime voicemail
On 12:13, Tue 27 Jun 06, Douglas Garstang wrote:
-Original Message-
From: Douglas Garstang
Sent: Tuesday, June 27, 2006 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Realtime Voicemail
I'm noticing that the documentation
HERE IS answer [EMAIL PROTECTED]had same problem..make the settings for 90 volt.. not 70 volt ringer.. make it trapezoidal not sinusoisalmake it 900 ohm not 600 impedence..that worked for pap2's
seem siemens are made for europe style ring voltage not north american.On 6/27/06, Herchi Silviu
-Original Message-
From: Michiel van Baak [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 27, 2006 12:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Realtime Voicemail
On 12:13, Tue 27 Jun 06, Douglas Garstang wrote:
-Original Message-
From:
We use cisco 7960's but thats not cheap..BTW Doungyour signature :Those that sacrifice essential liberty to obtain a little temporary safetydeserve neither liberty nor safety. -- Ben Franklin (1759)
is a good one.. tell that to your president..and the patriot act.s/patriot/cutallrights/PS Andrew..
A few days ago, I started getting these errors on my Asterisk (1.2.9.1) console:
-- Executing Queue(Zap/1-1, sales|tT|||3600) in new stack
-- Channel 0/2, span 1 got hangup
-- Channel 0/1, span 1 got hangup request
Jun 27 10:53:27 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring
So I've got a 601 (1.6.6) with the side car, and the buddy watch seems to be working but it updates the statuses unreliably. When I do a sip show subscriptions in asterisk it lists my phone 12 times and at the bottom it says 0 active SIP subscriptions(s) I've got an older CVS-HEAD build, pre
1.2,
-Original Message-
From: Douglas Garstang
Sent: Tuesday, June 27, 2006 12:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Realtime Voicemail
-Original Message-
From: Douglas Garstang
Sent: Tuesday, June 27, 2006 11:55
On 12:26, Tue 27 Jun 06, Douglas Garstang wrote:
I wasn't aware that realtime voicemail supported caching. I knew sip.conf
did, but voicemail?
How does that work?
I just tried setting 'format' and 'sendvoicemail' in the users database row.
No effect.
BUT... maxmsg DOES work... I don't
BLAH=1BLAH=1On 6/27/06, Brian Capouch [EMAIL PROTECTED] wrote:
Jon Schøpzinsky wrote: Hello As far as ive understood, you can just write
Exten = s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and ${AVAILSTATUS} would return 1Through more testing, the double quotes
I've
never seen that problem, and I've only ever used 1.2+ with Polycom and
buddies.
-Original Message-From: Ryan Stark
[mailto:[EMAIL PROTECTED]Sent: Tuesday, June 27, 2006 12:31
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users]
Sure... I would do the following 1. Set qualify =yesBash Script (in a cron) that doesa. asterisk -rx sip show peers foob. grep UNREACHABLE foo | wc -l mime-construct if output of the grep 1
hope this helpsrajeevOn 6/26/06, Roger Workman [EMAIL PROTECTED] wrote:
Is there a way to get asterisk to
this problem seems to occur in 1.2.9.1 (1.2.9 also? dunno about 1.2.8)
with users of chan_agent and agents making transfers. Kevin P. Fleming
[EMAIL PROTECTED] was looking at the issue last i read on this list.
check out the thread 1.2.9.1 crashed today on this list over the last
~1.5
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