On 16:38, Tue 01 Aug 06, Douglas Garstang wrote:
I suggest you use an AGI for it.
That gives you way more options
How does AGI help? Your still calling DUNDILOOKUP inside the AGI script, and
not matter how many times you call it, your still always going to get the
lowest priority path
For the VoIP phone question, I can warmly
recommend the Snom 360.
When using hints in asterisk, this is the
perfect phone for secretary use, as you can also add a side panel with 48 extra
buttons with lights.
When using hints you can see when
extensions a talking, ringing, as well as
Hi!
I'm having an interesting problem with Cisco7970 SIP load (8.0(2)SR1) -
the phone seems to work otherwise fine, but I can't do an assisted
transfer (and the 7970 phone also doesn't seem to support the BlindXFer
option that previous models have had). Phones are connected to Asterisk
Hi FriendsI want to configure my asterisk server with audio codec model MP-108FXS my sip.conf has the user name with mohit i want to configure this withanalouge SHOBHIT NIRALA CONT NO. 9871476403
Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail
--- Kai Ober [EMAIL PROTECTED] a écrit :
Khaled Chehab schrieb:
Is SRTP available in asterisk? Or how to
implement it ? am using trixbox
you asked this question before, and you got answers,
read your mail, or
stay away from this list!
Hi :)
I have a 'slow dialing' problem. When I dial 200# for the
'echo test' application from my PBX extension 1010, I see this
in the console the instant I press the # key:
-- Starting simple switch on 'Zap/65-1'
-- Accepting overlap call from '1010' to '200' on channel 0/3, span 3
so
I think I'm using native since I don't recall installing anything else (except
lame codec). How do I check which I am using? I'm unfortunately no asterisk
expert that's why I need your help! ;)
My musiconhold.conf (I have no musiconhold_additional.conf):
;
; Music on hold class definitions
;
Hi
just creat accounts and configure MP-108FXS
per port basis
its working in my setup
ram
On 8/2/06, Mr shobhit nirala [EMAIL PROTECTED] wrote:
Hi Friends
I want to configure my asterisk server with audio codec model MP-108FXS
my sip.conf has the user name with mohit i want to configure
Rxfax has no ECM, try hylafax and iaxmodem.
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paradise
Dove
Sent: 01 August 2006 21:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] rx_fax problem
hi,
rx_fax
I'm thinking this could be a queue problem?
But I still don't understand why the hell it just flips out after a few hours.
Now it all ran for about 12 hours since last reboot (longest so far). And this
config worked on my old install of asterisk...
Problem description (one of them):
Incoming
Yes this is possible, you just setup the softphones and maybe the provider in sip.conf and write your dialplan :)On 8/1/06, J Rangi
[EMAIL PROTECTED] wrote:Hello,Is is possible to setup an asterisk server with out buying Digium card.
I mean can we do this type of setup.We all know that X-Lite can
Hi
can anyone suggest the best way to go about installing Asterisk for a small
business of 20 - ive read all the documentation but the real issue is the
dial plan. Ive attempted to write my own which is similar to FreePBX's
dailplan. Ive even attempted to rewrite dialparties.pl simply to
Barzilai Spinak wrote:
Thank you Steve.
About the configs in Asterisk... I confess that I'm new to the code so
I still need to read more. I didn't know about ast_config()
About the hardcodedness of the countries... that seems to be the
problem. Everything is too oriented to my country works
Hello all,I'm a user of the latest version of TrixBox. I would like to know from the users if any one has implemented the follow on calling system on DISA.If you dont understand what i mean, let me make it clear.
When i'm calling from PSTN to my asterisk using DISA and call a trunk that i have
Hello list,
Howto be shure if one FXO module on TDM400P is not working because is burn out
or something like physically demaged. It worked for an almost a year then just
stopped. The next FXO module on the same card is working like charm.
Thanks.
Hi,
Try to swap both fxo modules.
This way you will notice if the module is out or another problem is present.
The tdm400 could be damaged or there is a configuration problem.
Regards,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Hello list,
Howto be shure if one
On Wednesday 02 August 2006 08:52, Rostislav Bagrov wrote:
Howto be shure if one FXO module on TDM400P is not working because is burn
out or something like physically demaged. It worked for an almost a year
then just stopped. The next FXO module on the same card is working like
charm.
Your
Hi Steve,
I need to enable the Unicall channel in my asterisk
box to be able to interconnect to a local telco
provider using MFCR2. I use the unicall release
unicall-0.0.3pre9 and a patch for asterisk 1.2.
Compilation was done with ease. The problem is that I
got an error Unable to read
hey folks
hope some one came across this problem
one of our polycom's just crashed
after reboot it comes up with this error
error loading 0004f204fcc.cfg
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Hi
all,
Anybody have experience (good/bad) with Dell Poweredge 1950 / 2950
?
I'll install
Fedora Core 5 or #PoundKey and Digium Hardware.
Regards
Fred
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Hello,
How can I build extensions.conf so that Asterisk routes calls based
on the ANI, not the number dialed.
Example:
All calls coming down a PRI are going to the same number. I would
like to route them to a new number based on the Calling-Station-Id.
I.E. All calls from
Hi.I've installed Asterisk with a MD3200 modem,zaptel modules recognize the card,when i dial to asterisk, it answers but when I Playback(something) do not receive any audio, only a sound like audio static
but I created in extensions.conf[demo]iclude= defaultand when in the console type the
Make sure you can access the file on your FPT server. Also make sure
that you did not fry the Ethernet port(s) on the phone.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stas Khromoy
Sent: Wednesday, August 02, 2006 9:50 AM
To:
i am afraid the second could be the case
since the whole block where the office is lost power yesterday
thanks
PS : FPT = FTP ? :)
Original Message
Subject: Re:[asterisk-users] polycom soundstation 501 crash
From: Alexander Lopez [EMAIL PROTECTED]
To: [EMAIL PROTECTED],
Is possible that you are missing the XML file with the supertones
definitions. Usually is located at /usr/share/spandsp/global-tones.xml
, but it depends on how you configured the spandsp package
(./configure --prefix=/usr/blah). Notice that spandsp and
libsupertone should be configured with
This means the phone is attempting to load this configuration file and cannot find it from your boot server. The phone at this point must have a boot server with these files. Put this file on a FTP server and point the phone to that server to pickup and download this file. Jessee HolmesAtacomm /
not to sound like an idiot
but where do i get the files ?
these guys ?
http://www.polycom.com/resource_center/0,1454,pw-6812-12612,FF.html
SoundPoint IP/SoundStation IP SIP Software 1.6.7
SoundPoint IP/SoundStation IP BootROM 3.2.1
Original Message
Subject:
I think still didnt explain me clearly
The problem is when I dial 0, in this case the asterisk
take Zap (connected directly to ext 200 from Panasonic), Panasonic gives tone,
dial another extension (ie 100), the extension rings but when answer the phone asterisk
keeps ringing it doesnt
Swap the modules and see if fault moves with the module. If it moves then
the module could be faulty, if not, the module is ok
neil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rostislav
Bagrov
Sent: 02 August 2006 13:52
To:
Hi all,
I have a problem sending attached files with voicemail. I have postfix
installed.
When I write attach=no in voicmail.conf the notification is sent with
no problem. But when I change to attach=yes, the notification never
arrives.
Could it be a postfix problem? Anybody could tell me how
At first, you would have to get these from your service provider or your reseller. They should have them available. I wish I could find a sample of one of the .cfg files, but I can't seem to locate it at this moment; however, here is a starting sample from Polycom.File Name: .cfg?xml
Hi,
I have been looking through all the web sites about echo problems and how to
solve them on the spa-3000, but I still have not managed to fix mine! I'm in
the UK and have setup all the tones, port impedance to 370+620||310nF, had
the echo Canc options on and off, turned down the SPA - PSTN
Is their a way of implementing arrays in asterisk?? What im trying to do is
allow the user to input numbers (internal and external) from his line thats
gets saved via astdb - (follow me purposes) - i would like to look t each
number individually and check for CF, CFB CFU etc etc . can this be
error loading 0004f204fcc.cfg
Missing a digit?
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Hello everybody,
I'm having a problem trying to dial with an IAX2 extensions. I connect
trough iaxComm and try to dial an extensions, then in asterisk CLI
appears this:
Aug 3 01:14:29 NOTICE[20915]: chan_iax2.c:7357 socket_read: Rejected
connect attempt from 192.168.1.128, requested/capability
I've posted on GAF (Free Lance Site) a request for bids for
modifications to Asterisk PBX source.
If you are interest in bidding on this, please view it at
http://www.getafreelancer.com/projects/78138.html
Thanks you for your time.
Bart Fisher
[EMAIL PROTECTED]
It's solved. The problem was that the softphone has only one codec
allowed and asterisk was configured to no allow that codec.
On 8/2/06, Facundo Ameal [EMAIL PROTECTED] wrote:
Hello everybody,
I'm having a problem trying to dial with an IAX2 extensions. I connect
trough iaxComm and try to dial
Rostislav,We just experienced this ourselves. First our second module stopped working, then our fourth. We popped an old Sangoma card in and the lines worked fine, so I figure it must be the modules. I'm going to swap the modules in a tester box soon and see if the suspected faulty ones light up.
Dean @ INKnBITs wrote:
Hi,
I have been looking through all the web sites about echo problems and how to
solve them on the spa-3000, but I still have not managed to fix mine! I'm in
the UK and have setup all the tones, port impedance to 370+620||310nF, had
the echo Canc options on and off,
On Wednesday 02 August 2006 18:09, Pele Zico wrote:
Is their a way of implementing arrays in asterisk?? What im trying to do
is allow the user to input numbers (internal and external) from his line
thats gets saved via astdb - (follow me purposes) - i would like to look t
each number
Hi,
Having checked the documentation for SIP_HEADER:
pitux-exercice15*CLI
-= Info about function 'SIP_HEADER' =-
[Syntax]
SIP_HEADER(name)
[Synopsis]
Gets or sets the specified SIP header
I thought I could write some info in SIP_HEADER to retrieve them later.
But when I try to write to
I'm about ready to give up on IAX2. It seems to have some SERIOUS limitations.
Incoming PSTN call comes into to user A on pbx1. We look for user A locally,
and don't find them. We then do a DUNDi lookup, get a path, and dial user A on
pbx2 with IAX2. User A picks up the call.
When IAX passes
Hey guys,I'm having yet another strange problem. I've recently set canreinvite=yes, allowing the RTP streams to avoid our * server. Now, a few people are experience one way audio drops on internal calls. External calls are working fine (they re-invite directly to a Cisco router). Sometimes, if you
Anyone seen DTMF control lost intermittently inside a menu?
We have had a few problems with one particular menu but both seem identical but
maybe different traffic.
Shane
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Then you have something wrong some other place, if you are using an
FXO card then asterisk is not even giving you the ring, the panasonic
is.
On 8/2/06, Pablo Mora [EMAIL PROTECTED] wrote:
I think still didn't explain me clearly…
The problem is when I dial 0, in this case the asterisk
Hi,
I'm currently running Asterisk 1.2.10 on a dual Xeon 3.4GHz with 2GB RAM
and SATA drives, but I'm only able to achieve 120 calls with good audio
quality (using G.711u). I'm using realtime for voicemail accounts and
ODBC for voicemail storage along with one MySQL when dialing out. The
I have some ethernet cable splitters I'm not using any more. They go in
pairs, one plugs into the wall socket in the office, the other plugs
into the other end of the same cable in the server room. each gives two
female ethernet sockets that represent two separate network cables, each
using
On Wednesday, August 02, 2006 6:49 AM Eric ManxPower Wieling wrote:
Zap/10-43 would indicate that this is the 43rd call (call waiting) on
channel 10. Obviously this would have to be removed to do it the way
you want.
Obviously. :-)
Or we find another solution for the problem/challange...
- Original Message -
From: Gary Richardson
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 13:54:04 -0300
Subject: [asterisk-users] canreinvite=yes
and RTP dropping in and out
Hey guys,
I'm having
Out of curiosity - are you running an smp kernel or a uniproc kernel? I am doing some benchmarking as well on a similar system (dual XEON 3Ghz with 4GB RAM and SATA drives in mirrors) and am seeing the uniproc kernel performing better under CentOS
4.3 (2.6.9-34.0.2.EL) when testing general server
I was wondering if we could uplink small switches to the wall data ports
to
the switch, and connect the additional SIP phones to them to get them
connectivity to Asterisk?
Yes, we do it and it works fine, as long as you don't cascade more than 3
switches between two devices your latency
My next attempt at this is going to be putting a hub in between the path to the switch. I'm hoping to be able to sniff the packets to see what's going on.Also, using the network status page on the hard phones, the transmit and receive counters for the direction of the channel slows way down as if
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 13:50:56 -0300
Subject: [asterisk-users] Limitations of
IAX
I'm about ready to give up on IAX2. It seems to
With Asterisk 1.2.* and TRUNK, I've noticed some odd behavior with SIP registrations and connectivity over the past day. First, I noticed Asterisk REFUSED to register any trunks over SIP, prompting a lot of timeout messages. It also refused to accept registration requests from internal phones,
DIALEDPEERNUMBER contains the exact peer spec for the peer that picked up.
You can use that.
From: [EMAIL PROTECTED] on behalf of Koopmann, Jan-Peter
Sent: Wed 8/2/2006 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Even if he has r in the dial plan?
Jorge
C F wrote:
Then you have something wrong some other place, if you are using an
FXO card then asterisk is not even giving you the ring, the panasonic
is.
On 8/2/06, Pablo Mora [EMAIL PROTECTED] wrote:
I think still didn't explain me clearly…
Hi All
I have a client with 3 analogue gsm routers, one of which is a 'Fusion
100'. The other 2 routers work perfectly, but the 'Fusion 100' router
refuses to dial. I can dial from an analogue phone connected to the
router.
From the CLI in debug mode, I can see polarity switches when trying
Hello, I hired a consultant to setup and asterisk box for me.
I am trying to learn how to maintain some of the things myself,
because the response time on maint requests from this wonderful
consult are brutal (u know once they have their money)
anyways. currently we have it set up so when
Benchev wrote:
On Wednesday 02 August 2006 18:09, Pele Zico wrote:
Is their a way of implementing arrays in asterisk?? What im trying to do
is allow the user to input numbers (internal and external) from his line
thats gets saved via astdb - (follow me purposes) - i would like to look
t
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 7:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Limitations of IAX
- Original Message -
From: Douglas Garstang
- Original Message -
From: Gary Richardson
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 14:34:31 -0300
Subject: Re: [asterisk-users]
canreinvite=yes and RTP dropping in and out
My next attempt at
On Wednesday 02 August 2006 09:42, Joshua Colp wrote:
As previously pointed out transporting the account code may be a security
risk as well. If you really need it why don't you encode it into the dialed
number or something?
Because that's a completely shitty workaround for something that may
Ok... it'd be great if someone could explain this to me...
User A on pbx1 wants to dial User B on pbx2. We do a local lookup and don't
find user B on pbx1, so we do a DUNDi lookup of user B, get a result, and place
the call to user B on pbx2 with IAX2.
When pbx2 calls the AGI script that
On Wednesday, August 02, 2006 7:39 PM Vadim Berezniker wrote:
DIALEDPEERNUMBER contains the exact peer spec for the peer that
picked up. You can use that.
Consider yourself my hero of the day! That looks VERY promising. It does not
show the technology so
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 14:51:10 -0300
Subject: RE: [asterisk-users] Limitations of
IAX
How many CLEC's are you aware of that are
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 11:54 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Limitations of IAX
On Wednesday 02 August 2006 09:42, Joshua Colp wrote:
As previously pointed out
Joshua,
Thank you!! I didn't even notice that. I'll fix it and report the bug to
FreePBX.
Keith
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
Colp
Sent: Monday, July 31, 2006 11:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
No idea, but DIALEDPEERNAME should contain the same value as BRIDGEPEER. Try
that.
The only difference is that BRIDGEPEER is set slightly later (when the call is
bridged).
From: [EMAIL PROTECTED] on behalf of Koopmann, Jan-Peter
Sent: Wed 8/2/2006 2:05 PM
To:
Are there any credit card processing scripts for asterisk, that would
allow me to enter credit card number amount and dial my IVR system?
--
#Joseph
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On Wednesday 02 August 2006 10:29, Joshua Colp wrote:
Writing it directly into the protocol is the dangerous part. You would need
some control over who can set it and under what circumstances it is
allowed. By encoding it into the dialed number only the users you control
will be able to do it
Quoting Joseph [EMAIL PROTECTED]:
Are there any credit card processing scripts for asterisk, that would
allow me to enter credit card number amount and dial my IVR system?
have a look at www.opayc.com - while not specifically for asterisk, these
drivers use odbc (on unix or windows) to talk
I've trying to use DUNDi with SIP to see if it works around some limitations of
IAX2.
I do a DUNDi lookup, get my SIP path, and try to dial it. Asterisk immediately
says 'No such host', eventhough that's the path is just returned!
[Aug 2 13:07:05] == Spawn extension (global_vmdeposit,
I don't need a gateway; I was looking to find a script what would let me
dial into our IVR system, provide merchant number + device number +
credit card + exp. date + amount
Merchant #, device # are constant so it can be build into the script;
credit card #, exp. date and amount are variable so it
Using the SECRET variable for sip doesn't work.
On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote:
I've trying to use DUNDi with SIP to see if it works around some limitations
of IAX2.
I do a DUNDi lookup, get my SIP path, and try to dial it. Asterisk
immediately says 'No such
You need to carefully consider outside VoIPproviders
IMHO. I would look for providers who are very upfront about their network
architecture and how they connect to the PSTN (the public telephone
network). As a minimum, I would ask for IP addresses to some of their SIP
servers and check ping
Join the club,
I would like shared line appearance ability along with MANY
other people too. That feature is only now on the rader screen for
Asterisk but it probably won't be available for some time. Months if not
years (probably not till v1.6 which is after the next version which will be
Quoting Joseph [EMAIL PROTECTED]:
I don't need a gateway; I was looking to find a script what would let me
dial into our IVR system, provide merchant number + device number +
credit card + exp. date + amount
Merchant #, device # are constant so it can be build into the script;
credit card #,
I think what the poster meant was that this script would conceivably
need to be designed to work with the API of a SPECIFIC gateway and might
not be easily generic.
Joseph wrote:
I don't need a gateway; I was looking to find a script what would let me
dial into our IVR system, provide
Secret? Do you mean sbsecret in sip.conf?
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 1:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi with SIP
Using the SECRET variable
- Ability for the phone to ring when the receptionist is on one call
and a second or third call is incoming. (this has been the biggest
frustration up to now. When a second call comes, there is no tone
that heard on the IP500. Perhaps I am missing a setting?)
The Snom 360 can
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
I've trying to use DUNDi with SIP to see if it works around some limitations
of IAX2.
I do a DUNDi lookup, get my SIP path, and try to dial it. Asterisk
immediately says 'No such
host', eventhough that's the path is
I'm talking about the rotating DUNDi secret that is stored in dbsecret
in iax.conf. It doesn't exist in the SIP channel.
On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote:
Secret? Do you mean sbsecret in sip.conf?
-Original Message-
From: Aaron Daniel [mailto:[EMAIL
unsubscribe
Keith HerringtonTechnical Support EngineerTwisted Pair
Solutions, Inc.Main Support: +1 (206) 812-2390Direct: +1 (206)
812-2375Cell: +1 (206) 427-5285Fax: +1 (206) 812-0737This
transmission and any files attached to it may contain confidential and/or
privileged information and
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 2:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: DUNDi with SIP
In article
[EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
I've trying
So what are the options?
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] DUNDi with SIP
I'm talking about the rotating DUNDi secret
Hi all,
has anybody any experience with Ateus Easy Gate connected via Digium card to
asterisk? It works fine for me except it doesn't pass the caller id and the
hangup detection is quite slow. Are there some tips how to shorten the hangup
delay?
Thanks.
Jan Fousek
You can use an unchanging password. It's not as secure, but it will provide functionality.AlexOn 8/2/06, Douglas Garstang
[EMAIL PROTECTED] wrote:So what are the options? -Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]] Sent: Wednesday, August 02, 2006 2:03 PM To:
I've tried doing it without a username/password as described at:
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
but then authentication to the INVITE fails. I'm authenticating on the from:
field, ie the individual user, which I don't think is right.
I've
Alex,
Thanks... I haven't had any luck with it yet.
My
dundi.conf has:
180netsip =
global_dundi_local,1,SIP,dundisip:[EMAIL PROTECTED]/${NUMBER},nopartial
and my
sip.conf has:
[dundisip]type=usercontext=global_dundi_localsecret=password
A
DUNDI lookup on the console returns a SIP
On Wed, 2006-08-02 at 15:33 -0400, Jon Pounder wrote:
Quoting Joseph [EMAIL PROTECTED]:
I don't need a gateway; I was looking to find a script what would let me
dial into our IVR system, provide merchant number + device number +
credit card + exp. date + amount
Merchant #, device # are
Doug,Two things: If you try to place that call manually (either via dialling it from a phone that supports SP URIs or by making an ext. for it in your dialplan and calling that extension), does it work properly? Are you able to place the call? If not, is the CLI output the same as when you try it
Alex,
Yep, I
can dial 9220370 directly. I have two extensions on pbx1 and two on pbx2. I can
place calls from 9220371 to 9220370 which goes through pbx2 only, and all is ok.
9220370 and 9220371 are registered on pbx2.
I had
this all working with IAX. I didn't change the keys... so I would
Hi,
I'm recieving the following error in my asterisk log (when starting *):
chan_zap.c: Failed to read gains: Invalid argument
Why? Attaching my zapata.conf and zaptel.conf. Using TE405P.
Thanks!
zaptel.conf:
span=1,1,0,ccs,hdb3
Hi,
Can you give a quick example on how to query an EXTERNAL database?
Thank you.
Andy
On 7/29/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Jul 28, 2006 at 04:08:19PM -0500, shawn bright wrote:
i would use a dial plan, but we are monitoring about 1200 units in the
field, i thought a
Thanks for the information.
I have been a paying a lot to make international calls. After reading
through this mailing list I have a feeling that a system can be setup
easily where the telecommunication can be affordable. Still remember
those days when I had to wait for two weeks to talk to my
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 2:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: DUNDi with SIP
In article
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 3:49 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: DUNDi with SIP
In article
[EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 3:49 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: DUNDi with SIP
In article
[EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Lo there, i have an app that needs to initiate a phone call on a zap channel.i have been able to test it out ok with the method of dropping a call fileinto the /var/spool/asterisk/outgoing and specifing the phone number in the call file.
what i need to do, however, is initiate a phone call from a
The way to make this work is to define a sip user/peer with the IP
address in it, then have your dundi.conf entry look like:
180netsip = global_dundi_local,1,SIP/peername/${NUMBER},nopartial
As far as I can tell from the code, this is the only way to make it work
properly based on the way the
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