On Tue, 8 Mar 2005, Tom Samplonius wrote:
On Tue, 8 Mar 2005 13:36:39 -0700, Dr. Matthew Roller
[EMAIL PROTECTED] wrote:
When I forward my PSTN phone(Qwest) to my cellphone and someone calls
it, my cellphone(ATT) shows an arrow next to the caller id showing it
is a forwarded call, is
Jim Van Meggelen [EMAIL PROTECTED] writes:
You could do that with two tin cans and a string! ;-P
...so, the next time you want to complain about your phone service,
why don't you try using two Dixie cups and a string? We don't care.
We don't have to. We're the Phone Company.
--Lily
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this response
(on call out). Any suggestions? I don't think it is a problem with the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my
On Wed, Mar 09, 2005 at 06:38:24PM +1100, Callum McGillivray wrote:
Hey all,
Hi, welcome to this list
My apologies if this sounds blindingly obvious, but am I correct in saying
that I can use Asterisk to connect two extensions and make calls between
them without needing an actual telephone
On Wed, Mar 09, 2005 at 02:47:29PM +1100, Mike Sander wrote:
This is a re-post as it was pointed out that I replied to a different
thread instead of creating a new post. Sorry for the additional traffic.
Mike
Dear All,
I understand the excitement surrounding a service like Asterisk, and
Leo Ann Boon wrote:
Another question... Are you aware of a SIP ATA or phone that has some
kind of VPN (i.e. PPTP) client embedded in? This would make the NAT
problem go away nicely and provide added security...
The Zulty's phones support VPN. Then again, many firewalls don't pass
through VPN
Hi
On Tue, Mar 08, 2005 at 11:53:14AM -0800, Victoria Alexandru wrote:
[snip]
Checking out from CVS:
[EMAIL PROTECTED] victoria]# cd /usr/src
[EMAIL PROTECTED] src]# export
CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
[EMAIL PROTECTED] src]# cvs login
Logging in to
:pserver:[EMAIL
On Tue, 2005-03-08 at 14:28 -0800, Victoria Alexandru wrote:
I'm not registered with wiki, but I can tell what was
the mod:
In rhconfig.h, in line 43 you'll find . I'll
try to email Mandrake people to have certitude but for
now what I did was to remove one pair of . I believe
this is a
hi all asterisk user
can you help me to find the way for hangup any call by pressing any key like
** or ## in astcc and place another call without providing calling card
number.
bashir
i search google to find out a
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To:
Thanks Vamsi
I have not been able to locate pwlib - 1.6.6 or openh323 1_13_5.
I found the latest versions through sourceforge and I found some older
versions on another site, but not these versions. This has been quite
frustrating. Anyway, I think by using the asterisk-oh323 branch under
Hi all,
I am able to receive voicemail in my mail box but when I try to play the
audio file attachment, I hear nothing at all (yet the caller on the other
end does leave a voicemail message)!
Anyone had a similar problem before? Ideas are welcome!
Note: I am using [EMAIL PROTECTED] 0.6
Thanks
On Tue, 2005-03-08 at 14:16 -0500, John Fullington wrote:
I set up a monitoring system that calls my techs when a problem occurs on
one of our networks, everything works fine unless asterisk calls a cell
phone in which case the tech can not respond using dtmf. It works fine if
the tech call
I am newest to this group and would appreciate your
help!
Is it possible to use quicknet phone jack with
[EMAIL PROTECTED] ver 0.6? Little
has been mentioned about use of quicknet products'
adaptability with
[EMAIL PROTECTED] I do have a couple of old jacks to
startup right away. Your
guide is
thanks for replying but no change at all any other tips,suggestions
thanks in advance
On Wed, 9 Mar 2005 01:44:41 -0600, Jay Milk [EMAIL PROTECTED] wrote:
You'll need canreinvite=no to each sip section in sip.conf, if you want
* to stay in the loop.
-Original Message-
From: Adnan
hello
i am using asterisk-oh323-0.7.1. i want to convert sip
call to h323 (h323 sjphone or h323 proxy). what could
be the best way for this. i am successfull in
converting h323-sip by using asterisk as gateway.
help required on sip-h323.
kamran
Is it true music on hold isnt supported
in IAX/2? I check the docs and it doesnt show a configuration setting in
IAX.conf and when I put someone on hold they dont hear the music and *
doesnt start the music on hold. If it doesnt is there a way to make this
Hello all,
It 's dificult to explain; The system I need is an box option (based on *),
that I would add to an existing PABX (ie: Nortel with 600 ext).
I need two E1/T2 card to plug the system between Telco (FT) and PABX (Nortel)!
One card for France Telecom Side (E1a) and one other to Nortel
Hi all,
I'm researching random call drops on our Asterisk and would like to
make sure whether it's something wrong with our VoIP provider or with
the Asterisk. I sniffed traffic between Asterisk and our VoIP
provider's SIP gateway, and observed that in the middle of the
conversation an RTP
On March 9, 2005 03:49 am, Tzafrir Cohen wrote:
The wiki has a section of exammple setups and configurations. What is
the atvantage of your separate site?
Please take that as constructive critisism.
The wiki is very messy and hard to find information. And I say this as an
experienced
On Wed, Mar 09, 2005 at 01:51:09PM +0200, Dipole Moment wrote:
I'm researching random call drops on our Asterisk and would like to
make sure whether it's something wrong with our VoIP provider or with
the Asterisk. I sniffed traffic between Asterisk and our VoIP
provider's SIP gateway, and
Hello,
I am trying to make a call from our PABX to Asterisk on PRI interface.
How can iconfigure Asterisk to enter the overlap receiving state if the complete number is not obtained in setup message.
Looking forward to any help in this regard
Regards
Nauman Bin
1.4.1 over here.
Jerry wrote:
Never had any of my 100 or so act like that. What version of code are
you running? I think 1.4.1 is the latest.
On Mar 8, 2005, at 8:05 PM, Ben Ruset wrote:
Hello list:
I have a very odd problem. Seemingly randomly, my Polycom IP600 phones
will ring without a call
On Tue, 08 Mar 2005 17:34:31 -0500, Jerry Geis [EMAIL PROTECTED] wrote:
SEPDEFAULT.cnf is not required nor recommended. The 7912 only uses the
gkMAC file
and the software version CP7912XXX file
The gk file must be lower case..
This phone 192.168.255.250 is requesting SEPXXX
It is
I'm sure this is a stupid question, but I'm not finding an answer
anywhere. Do I need a dedicated box to run asterisk, or can I put in my
server (running Fedora) and leverage some of the free cpu cycles and
disk space? Thanks,
Dunc
___
On March 9, 2005 07:26 am, n a wrote:
How can i configure Asterisk to enter the overlap receiving state if the
complete number is not obtained in setup message.
I take it the overlapdial=yes option isn't doing what you want?
Perhaps a more detailed explanation of what you're after would help,
Dunc,
Depends on the environment you run it in.
If this is main telephony system for a business, then a dedicated
machine is highly desirable, and you may also want to think about
redundancy and failover.
If it's for your own personal use, or it's a development machine, then
it can co-exist
Separate box is best.
If you are only new to asterisk go and download [EMAIL PROTECTED]
http://asteriskathome.sourceforge.net/
It's a iso you can download that does all of the configuring and setup
for you automatically.
Cheers
dean
-Original Message-
From: [EMAIL PROTECTED]
On March 9, 2005 07:31 am, asterisk wrote:
I'm sure this is a stupid question, but I'm not finding an answer
anywhere. Do I need a dedicated box to run asterisk, or can I put in my
server (running Fedora) and leverage some of the free cpu cycles and
disk space? Thanks,
That's a very
Just out of interest, has anyone tried Asterisk @Home on User Mode Linux?
Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
http://integrics.com/
dean collins wrote:
Separate box is best.
If you are only new to asterisk go and download [EMAIL
Hi John,
You didn't say what kind of cellular system. If its an AMPS system (I
don't think any other analogue cellular stiff exists) DTMF is quite
troublesome. If it is a digital network the DTMF actually comes from the
basestation, rather than the phone. Its is normally very high quality.
Why would you want to, it's a single iso, takes only 15 minutes to
install make your config changes for your particular machine and then
use the backup feature.
You bust anything irreparable, just load the iso again and load the
backup.
Up and running again in under 20 mins.
Cheers,
Dean
Dunno if I did make myself clear.
I want to route an incoming ISDN call using the excess digits dialed.
Need this for Fax.
If I understood your post and the wiki, using NV(Fax|Background)Detect
should Just Work, like in the example.
Has anybody done this with an ISDN line? Will, if the user
I am using asterisk with a handful of DM04B cards.
Everything seems fine except for an echo on all calls on the local end of the
call. In almost all cases the echo goes away after 15 to 20 seconds. I am
attributing the echo going away to the echo cancellation code that was enabled
when the
I am using asterisk with a handful of DM04B cards. Everything seems fine
except for an echo on
all calls on the local end of the call. In almost
all cases the echo goes away after 15 to 20 seconds. I am attributing the
echo going away to
the echo cancellation code that was enabled when
Dean,
- It's something new and fun to try!
- For testing clustering, advanced routing between Asterisks, etc,
without having to buy lots of machines.
- I'm not suggesting it at the minute as it's not proven, but perhaps at
some point in the future, offering multiple customers their own
Hello all,
I just installed a TDM400P with 2 FXO modules on my asterisk server. The
card works perfectly.
To get users to ring out from my SIP phones i setup an extension with 0 that
basically does something like this:
extension = 0,1,Dial(ZAP/g1) where g1 is the group of the two FXO channels
I have a linux (bash) script file which is invoked via:
exten = s,3,System(./BuildMsg.sh ${SFile} msg02 msg99)
When I start asterisk with the command: asterisk -gc
This script executes as expected ('asterisk -gc' and 'asterisk -vg' also
work). However, when I try to start asterisk with
Caller Name set using SetCIDName or SetCallerID is not displayed by
Zyxel P2000W (Firmware VWJ000F). The same problem has been
mentionend before, but I did not find any solution or hint.
http://lists.digium.com/pipermail/asterisk-users/2005-January/082801.html
--
Stefan Tichy [EMAIL
Hi List,
I'm using VoIPJet and NuFone as a fallback, and it seems that both of
them are circuit busy!
Also it seems that VoIPJet takes forever to return 'circuit busy' while
NuFone does it instantly.
At any rate, is there like a reliable third VoIP provider I can use for
fallback when the two
CHris,
I had the exact same problem with the exact same error.
My password was entered incorrectly in context section.
The register line had the correct password. That is why you get
incoming calls. and not outgoing.
Jerry
___
Asterisk-Users mailing list
I thought echotraining=400 was the default?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Wednesday, March 09, 2005 8:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Echo
Jason,
You are correct. The phone is brand new and running SCCP. The tftp server has
the upgrade info - the gkMAC file. the 3 phones are not picking it up. THe
other 5 phones did it just fine.
You are correct the phone needs upgrading. Th gkMAC file pointes to the
upgrade file. The 7912 is not
the following is on voipjet's site:
Please note we are having a temporary glitch with our New York
location. Please send traffic to our West Coast Premium Server until
the problem is fixed sometime today. New SERVER IP: 69.25.60.30
although i guess an email to this effect would have been nice.
Try changing the extension from Broadvoice1 to the actual phone number
(and don't send your secret in a public email or maybe that's Chris'):
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=XXX
username=8475100139
Steve,
The cellular system is Cingular, and as I said it works fine if the call is
made from the cell phone to asterisk, so I don't think it's the cell switch,
If the call is made through the asterisk box using a pri line, Digum T100P,
to a cell phone then the DTMF does not work, for any
On Wed, 2005-03-09 at 14:34 +0100, Fabrizio Mazzoni wrote:
Hello all,
I just installed a TDM400P with 2 FXO modules on my asterisk server. The
card works perfectly.
To get users to ring out from my SIP phones i setup an extension with 0 that
basically does something like this:
extension
Hello,
we are a firm who wants to develop some VOIP solutions.
The first infrastucture we choose for development is:
- an Asterisk machine connected to a traditional PBX (s0). In this way
people is not (yet) obligated to migrate its extisting PBX (and analog
phones) to VoIP.
- The PBX will be
1.4.1 over here.
Just to rule out all the possibilities - it's not MWI is it?
http://www.voip-info.org/tiki-index.php?
page=Getting%20MWI%20on%20Polycom%20Phones%20to%20work%20with%20Asterisk
It shows up as a little half-ring. It should also be accompanied by
the top LED flashing, and a
On Wednesday 09 March 2005 13:37, Jonathan Hobbs wrote:
I have a linux (bash) script file which is invoked via:
exten = s,3,System(./BuildMsg.sh ${SFile} msg02 msg99)
When I start asterisk with the command: asterisk -gc
This script executes as expected ('asterisk -gc' and 'asterisk
On March 9, 2005 08:41 am, Jean-Michel Hiver wrote:
I'm using VoIPJet and NuFone as a fallback, and it seems that both of
them are circuit busy!
What exactly is the return code for nufone? Your dialplan should look
something like this:
exten = whatever,1,Dial([EMAIL PROTECTED]/${EXTEN},,g)
At the moment I'm just trying to figure out how the technology works -
sort of proof of concept. Ideally, I'd like to move my business to such
an environment
at which time I'd definitely put the package on a dedicated box.
Dunc
Alistair Cunningham wrote:
Dunc,
Depends on the environment you
The WAV files attached to the emails are recorded at a very low volume
for whatever reason. Raise your volume higher and you'll hear them.
-Herman
Julius Kidubuka wrote:
Hi all,
I am able to receive voicemail in my mail box but when I try to play the
audio file attachment, I hear nothing at all
On Wed, Mar 09, 2005 at 08:37:02AM -0500, Jonathan Hobbs wrote:
I have a linux (bash) script file which is invoked via:
exten = s,3,System(./BuildMsg.sh ${SFile} msg02 msg99)
Don't assume the daemon runs in a certain directory. The working
directory of a daemon should generally be '/'
Hmmm, maybe the dtmfmode is incorrect. in your sip.conf what is dtmfmode set to?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hi Cameron,
Thanks for the suggestions. I think this is precisely what I was
looking for, unfortunately neither of those variables appears to be set
on my incoming calls. This is probably because I'm doing remote call
forwarding which is done by the phone company rather than regular call
You'd have to trace the code to work it out properly. But ICMP packets
aren't generally passed to userspace. What's more likely is that the
kernel, upon receiving sufficient of these errors, decides the
connection is dead and notifies asterisk.
That's what I'm thinking and just want to make
In data Tue, 8 Mar 2005 18:25:38 + (GMT), hai scritto:
[chan_zap.so]Mar 8 17:53:06 WARNING[2447]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/modules/chan_zap.so: undefined symbol:
ast_retrieve_call_to_death
Mar 8 17:53:06 WARNING[2447]: loader.c:391 load_modules: Loading module
I'm running * with a bog standard HFC ISDN card using zaphfc. Everything
seems to work, including incoming calls, but I simply cannot make outgoing
calls. This is very odd since the same card worked with the same
configuration in another server.
This is what I get from * debug. The only
Since yesterday the little iaxy does not register anymore!
It is also not pingable with the last know IP address!
Unplug plug in the power cable reacts in a short flashing of the
Ethernet port and after a second or two a short flash of a green LED.
What should I check next?
bye
Ronald
i am using gnugatekeeper. i have three things
gatekeeper ip, account, accountpassword how to set
account and password in oh323.conf
gatekeeper=gnu gatekeeper ip
gatekeeperPassword=accountpassword
accountCode=account
is this ok any example how to use this i want to rout
my sip call to this
On Wed, Mar 09, 2005 at 12:58:05PM +, Alistair Cunningham wrote:
Just out of interest, has anyone tried Asterisk @Home on User Mode Linux?
IIRC it should not work, as user-mode-linux requires a special kernel.
You can try to use its tarball .
Anyway, I regularily test Rapid installation on
On Tue, Mar 08, 2005 at 09:05:34PM -0500, Ben Ruset said:
Hello list:
I have a very odd problem. Seemingly randomly, my Polycom IP600 phones
will ring without a call being placed to it.
That is to say, a random phone will ring. Nothing shows up under Caller
ID. Even the buttons that
No. The phone itself physically rings as if it was getting a call. No
lights light up, nor does it show a missed call, nor does it show CID.
Noah Miller wrote:
1.4.1 over here.
Just to rule out all the possibilities - it's not MWI is it?
http://www.voip-info.org/tiki-index.php?
Thank you for the response. I still have the errors mentioned below, sip
response and Failed to authenticate on INVITE
[PP]
type=peer
username=PP
fromuser=PP
authuser=PP
fromdomain=sip.broadvoice.com
secret=XX
host=sip.broadvoice.com
dtmfmode=inband
They are all POE. Fed from a Cisco switch.
Walt Reed wrote:
You can run ethereal to capture all packets to / from the phone.
Something is obviously causing this problem. If nothing shows up in
ethereal, maybe there is a power problem. Are your phones POE or
wall-wart?
-Original Message-
-this is very true, however, the current version of the Axxess software
(9.0) supports SIP trunking natively on the IPRC. I just got my Axxess
upgraded and am salivating to get * connected to it.
Hmm, so 9.0 is out and it supports SIP natively. How did you plan
Have you tried this:
http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
Zanzamar Majere wrote:
Thank you for the response. I still have the errors mentioned below, sip
response and Failed to authenticate on INVITE
[PP]
type=peer
username=PP
fromuser=PP
Andrew Kohlsmith wrote:
On March 9, 2005 03:49 am, Tzafrir Cohen wrote:
The wiki has a section of exammple setups and configurations. What is
the atvantage of your separate site?
The wiki is very messy and hard to find information. And I say this as an
experienced Asterisk user (multiple PRI
Jean-Michel Hiver wrote:
Hi List,
I'm using VoIPJet and NuFone as a fallback, and it seems that both of
them are circuit busy!
How are you determining a fallback condition from one voip to another?
greg
___
Asterisk-Users mailing list
Disabled echo canceller because of tone (tx) on channel 10
I understand that the PSTN companies use their own echo canceller's,
send a tone across 2100hz, the problem we're having is people are
complaining of echo on random calls. I'm assuming this may be the cause.
Is their anyway to 'ignore'
Jerry-
Thank you
I accidently sent my password on the LISTSERV last night so I just
changed (pasted) the new one in.
Still the same problem...
Mar 9 09:51:13 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to 'Chris Nibeck
sip:[EMAIL PROTECTED];tag=as4b70f2e7'
On March 9, 2005 10:43 am, Cirelle Internet Products wrote:
How are you determining a fallback condition from one voip to another?
Mine's rather simple but it works well:
[macro-nufone-dial]
exten = s,1,GotoIf($[$ACCOUNTCODE != ],s,gotac)
exten = s,n,SetVar(ACCOUNTCODE=${ARG2})
exten =
Cisco 7960 Upgrading from 6.x to 7.3 get Protocol Invalid. I'm sure this
has been discussed but has anyone figured this out.
Regards,
Juan Staalenburg
Teksavers, Inc.
(512) 255-8395 x1002
AIM: juanteksavers
___
Asterisk-Users mailing list
Thanks MF,
Yes that was me that sent my PW :-) It is changed now.
Same error...
Mar 9 10:12:46 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to 'Chris Nibeck
sip:[EMAIL PROTECTED];tag=as0cefa74c'
Sip.conf...
[*8475100139*]
type=peer
;user=phone
Cisco 7960 Upgrading from 6.x to 7.3 get Protocol Invalid. I'm sure this
has been discussed but has anyone figured this out.
See the Wiki. It's all there for ya. Don't recall the exact page name.
Try searching on 7960 and brick.
... JG
--
Joe Greco - sol.net Network Services - Milwaukee,
Yeah. Edit zconfig.h and there's an option to ignore 2100hz. I didn't know what caused the 2100 until you said something.
On Wed, 2005-03-09 at 09:47, Matt Schulte wrote:
Disabled echo canceller because of tone (tx) on channel 10
I understand that the PSTN companies use their own echo
On Wed, Mar 09, 2005 at 09:29:20AM -0600, Don Pobanz wrote:
Andrew Kohlsmith wrote:
On March 9, 2005 03:49 am, Tzafrir Cohen wrote:
The wiki has a section of exammple setups and configurations. What is
the atvantage of your separate site?
The wiki is very messy and hard to find
there are two of us with the same problem so I will answer for me. Yes
I tried the below instructions.
The current thinking by multiple people is * never tries authenticating
so removing the FQDN will force * to go to the related section named by
either a phone number or a non Fully Qualified
Hi,
Does anyone know of a Print-to-Fax client
that works with asterisk spandsp? Astfax is a partial solution but
that only lets us email the fax in, we'ld like to set it up so the user
can hit the print button and send the fax (even if all it does is email
- transparently to the user - the fax
Am I the only one seeing problems with chan_sccp and the latest Asterisk
stable? Is there anyone where it is still working?
My phones disappear after half an hour and are seen as dead by *
Thx!
___
Asterisk-Users mailing list
Hi ml, I'm experiencing some problem detecting hangup with sip channel. I
have an asterisk on remote site behind NAT and two xlite at home behind nat.
I can make calls between them but hangup cannot be detected.
When I try to hangup a call I see xlite that tell me hanging up for some
seconds and
Try setting your dtmfmode=inband in your your sip/iax/zap configs.
This forces them to default to inband.
You can then overide with info or whatever you need in other contexts.
James
On Wed, 09 Mar 2005 12:53:02 +0200, Mark Elkins [EMAIL PROTECTED] wrote:
On Tue, 2005-03-08 at 14:16 -0500, John
I am using gnugatekeeper and asterisk.
My h323.conf:
[general]
bindaddr = ipaddress
tos=lowdelay
port = port
accountcode=xyz
gatekeeper = gatekeeper ipaddress
[xyz]
type=h323
prefix=123
context=default
extension.conf:
[default]
exten = _321,1,Dial(H323/${EXTEN:[EMAIL PROTECTED]
I've seen a fax-printer driver for Windows PCs in the source
(TurboPower's AsynchPro). Would be an interesting project to adapt it
for * use.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 09, 2005 10:31 AM
To:
Has anyone got this
to work? Under Idle Display Animation, the administrators guide says "For
example, a company logo could be displayed"..
In the ipmid.cfg
file, I enabled 'ind.idleDisplay.enabled' (ie changed it to 1), and under the IP
500 section, I added an entry for the bitmap that I
It is my understanding from monitoring lists and reading press releases
that on the BV BYOD plans, once you bump two paths (like a three-way
call), the next path is at the per-minute rate. However, since I can only
receive calls and can't seem to call out on BV, I can't test this...
James
Hi there,
I'd like to know if there's any way I can edit the fields asterisk
sends in an MGCP response to my devices, without having to mess with
the source code. What happens is that asterisk sends an F parameter in
an audit endpoint message I don't want it to send. Does anyone know I
can solve
Removing the quotes and eliminating s,3,gotoif did work but its not
what I am looking for.
What I want to do is the following: If a ani that comes in has 10
digits I want to change the ${CALLERIDNUM} to unknown. If the ani
is 10 digits just goto voicemail.
When I set up my [vmail] to look like
This configuration solved my problem. I could have sworn I tried this
before. I guess not. I did not need to apply the patch. Also, I am using a
regular Registration setup in my sip.conf not broadvoice's funky one...
The only thing I can surmise is that order of the variables matters.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Giorgio Mandolfo
Sent: Wednesday, March 09, 2005 8:59 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Which hardware for this solution?
Hello,
we are a firm who wants to
kurt x wrote:
[globals]
Setvar(DIGITS=10)
Try this instead...
[globals]
DIGITS=10
-Chris
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
The full path name fixed the problem with script execution.
Thanks for all the help!
Jonathan
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: March 9, 2005 9:16 AM
Subject: Re: [Asterisk-Users] Asterisk System() call error
On Wed,
TEXAN services are provided by special tariff for State agencies in Texas.
This is not available to the general public.
The numbers are FREE because the State spends $$$,$$$,$$$,$$$.00.
And this is only for the PLEXAR (enhanced centrex) type service.
There are Free DID numbers with some vendors -
To me it looks like the $LEN function is not working. When I do
verbose start to * I
see that it walks right through every step whether or not the ani is
10 digits or something else.
Would it be better to write an AGI script?
Kurt
On Wed, 09 Mar 2005 11:41:50 -0600, Chris Wade [EMAIL
The only reason i like this card, it is a cheap RTC which we need for conferencing. We also used ztdummy but Dell Poweredge servers we use, have ohci USB and ztdummy only works with uhci. I wish some one would come up with some other means to have RTC. :-(
On Tue, 2005-03-08 at 12:30 -0600,
Hi Guys,
Does anyone have knowledge about
max number of conference rooms, and max number of
conference callers in one room?
Thank you so much.
jintwo
__
Celebrate Yahoo!'s 10th Birthday!
Yahoo! Netrospective: 100 Moments of the Web
kurt x wrote:
To me it looks like the $LEN function is not working. When I do
verbose start to * I
see that it walks right through every step whether or not the ani is
10 digits or something else.
Would it be better to write an AGI script?
Kurt
I use LEN quite a lot, works perfectly. Go back
On Wed, 2005-03-09 at 11:31 -0500, [EMAIL PROTECTED] wrote:
Hi,
Does anyone know of a Print-to-Fax client that works with asterisk
spandsp? Astfax is a partial solution but that only lets us email the
fax in, we'ld like to set it up so the user can hit the print button
and send the fax
kurt x wrote:
[globals]
Setvar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} != $DIGITS}]?4:5)
exten = s,4,Setvar(${CALLERIDNUM}=Unknown)
exten = s,5,Voicemail(u${ext})
exten = s,6,Hangup
Oh, and it should be
exten =
We have purchased an Asterisk based PBX solution that is completely
setup with one exception, overhead paging. We have a powered Paging
System and want to find a way to set an extension (i.e. 999) to use the
sound card, which would in turn go out the paging system speakers.
I've seen several
1 - 100 of 223 matches
Mail list logo
