Good day,
does anyone have pulse dialing working ?
http://voip-info.org/tiki-index.php?page=Asterisk+zaptel+pulse+dialing
At the link above there is a statement:
configuration for European telephone lines will look like:
make_time=63
break_time=37
pause_time=800
So where these pamameters
Before hanging up, there should be an extension reminding everyone that
top posting is super duper wrong and oh so annoying.
Must I use the Wiki or Google to find out what top posting is? :-)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi.
I still hace problems getting my line to work. When I
start asterisk, I can see:
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, PRI Signalling signalling
...
-- Registered channel 31, PRI
bchan=1-15
dchan=16
bchan=17-31
Just a wild guess (I never worked with this equipment): try
bchan=1-15,17-31
dchan = 16
By the way: what is on channel 0 ?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Holger Schurig wrote:
Just a wild guess (I never worked with this equipment): try
bchan=1-15,17-31
dchan = 16
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
That's my setup and it works fine with our Teles switch.
By the way: what is on channel 0 ?
timing.
Apollon Koutlides
Hi list,
I installed asterisk with h323 support (on debian, from deb
ftp://debian.marlow.dk/ sid asterisk )
I tried, and have a quick look at the list archive and documentation.
But I could not manager to do what I want.
I want asterisk as a simple SIP = h323 Proxy Server.
SIP clients registers
Hi, you have to launch the script prepaid-make.sh
in the database directory and copy the prepaid.conf in /etc/asterisk.
reseaux [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
14/06/2004 18.24
Please respond to
[EMAIL PROTECTED]
To
[EMAIL PROTECTED]
cc
Subject
Re:
Its a matter of personal preference Holger, most people dont care, but the
ones who do whine about it a lot and can read it fine just as well. This is
a top post. its up top of the msg its posted about. It gives some the
impression youre too lazy to trim the stuff below it.
While trimming is a
Hello all,
This afternoon I had a BRI line installed by Telstra (our telco in
Australia). I'm using an Eicon Diva 2.02 PCI ISDN card with the isdn4linux
driver.
Incoming and outgoing calls with Asterisk work fine (and with no echo - my
main reason for getting ISDN). However, I can't seem to get
Thanks... I tried, but none of those solutions worked! :S
Any other idea?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hi all,
I'm stuggling with how to present calleds to a specific DDI (DID) with Music on hold
whilst the call is hunted around 3 phones, then if not answered within a certain
period forwarded to voicemail.
So far I've got the queue working and the voicemail but not both together.
Ive had a
Dear Sirs,
I've got a weird problem with IAX2 transfers.
My setup consist of 3 Asterisk servers. One is located in Europe on a
public IP and a local PSTN connection through ISDN. Two are located in
South-east Asia - both on public, but dynamic IP. These two each have a
bunch of SIP phones
Look at show application queue
It includes a timeout option, which after the call is not answered for
some period of time will drop back to the dialplan and hence your
voicemail.
Regards,
Adam
On Tue, 2004-06-15 at 18:35, Matt wrote:
Hi all,
I'm stuggling with how to present calleds to a
Further to my previous post;
It seems that if Asterisk recognises the DTMF digits, it will intercept them
and not send them to the ISDN card (either that, or the ISDN card isn't
regenerating them).
If I change the dtmf mode that the phone is using to something different to
what Asterisk is
This is an issues with DTMF clamping, you need to use chan_capi to get DTMF
working correctly.
Jason
At 18:30 15/06/2004 +1000, you wrote:
Hello all,
This afternoon I had a BRI line installed by Telstra (our telco in
Australia). I'm using an Eicon Diva 2.02 PCI ISDN card with the isdn4linux
Hmmm.
I tried:
exten = 7001,1,Answer
exten = 7001,2,Queue(test|t|||10)
exten = 7001,3,Hangup
exten = 7001,102,Voicemail(u100)
exten = 7001,103,Hangup
but that doesn't seem to work.
I'm sure I've made a stupidly obvious mistake; but I can't for the life of me see
what it is!
Cheers
Matt
Hi,
I'm trying to figure out what the issue is splicing Asterisk between our
Telewest PRI and a GDK-186 with a PRI card.
We're using the Digium TE405P
Our telco provider is Telewest, and Telco directly into switch is fine.
When I splice Asterisk in, I can make and receive calls from
What's that ?
Dial(SIP/-083601e0, ZAP/g1/h) ?
why 'h' ?
don't use exten = _.,1,blah , but
try with exten =_X.,1,blah
Matteo
Il ven, 2004-06-11 alle 23:59, Christian Gatti ha scritto:
ser forwards a sip message with extension 9996 to asterisk which
plays my 'userisoffline' message and
Hello,
I have a question about the configuration of the SIP telephone. The
situation is following:
We have two SIP telephones. One of them is configured to answer the
incoming calls from FXO or other directions. If there is no one
available to answer to the ringing phone, I would like to
Hi,
I'm pretty new to asterisk so excuse the stupid question:
what is the purpose of defining channels as trunks ?
I noticed that you can define Zap groups and IAX connections as trunk,
but the purpose is not clear to me ...
Tnx !
--
Best regards,
Alessio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jason Williams
Sent: Tuesday, 15 June 2004 6:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Outgoing DTMF when using BRI
i4l (Eicon Diva) - problems
This is an issues with DTMF
Steven Critchfield [EMAIL PROTECTED] wrote:
You forgot to add in how awful it is when people post using HTML and
then override font sizes or assume blue is an appropriate font color for
their message.
While I know some people don't like it when I turn my attention to them,
if it takes me
On Tue, Jun 15, 2004 at 07:43:32PM +1000, Shaun Ewing wrote:
That's the last thing I wanted to hear :-(
Apparently my ISDN card (Eicon Diva 2.02 as I mentioned) supports CAPI, but
I've only been able to find Windows drivers for it.
Have a look at http://www.melware.de
Maybe that helps.
--
Hi,
Checking while back, it was possible for two or more UA to use same
login info to place
calls and use other services at the same time!
Does anyone know if there were any development done in order to prevent
this?
___
Asterisk-Users mailing list
Dear,
Is there somebody who have experience with the Siemens Optipoint 400 standard SIP ?
And where can we buy it (i'm from belgium)
We are using for the moment Cisco 7960, 7905, snom 200, Mitel 5055 and in my opinion
the Mitel does his work the best combined with the 7905,
the 7960 is
Kevin Walsh wrote:
Steven Critchfield [EMAIL PROTECTED] wrote:
You forgot to add in how awful it is when people post using HTML and
then override font sizes or assume blue is an appropriate font color for
their message.
While I know some people don't like it when I turn my attention to them,
if
On Tuesday 15 June 2004 05:19, Matt wrote:
exten = 7001,1,Answer
exten = 7001,2,Queue(test|t|||10)
exten = 7001,3,Hangup
exten = 7001,102,Voicemail(u100)
exten = 7001,103,Hangup
I'm pretty sure you want
exten = 7001,103,Voicemail(u100)
exten = 7001,104,Hangup
The help for the Queue app
On Tuesday 15 June 2004 03:17, Holger Schurig wrote:
bchan=1-15
dchan=16
bchan=17-31
Just a wild guess (I never worked with this equipment): try
bchan=1-15,17-31
dchan = 16
By the way: what is on channel 0 ?
E1s start at channel 0?
I don't know anything
And didn't the original poster of this thread state rather forcefully that
this list is for * issues, not to be hijacked - which is exactly what is
happening based on comments/demands made by the original poster that were not
on the topic of *
Simon Brown
-Original Message-
From: [EMAIL
On Tuesday 15 June 2004 06:34, Chris Lee wrote:
Top posting is what a lot of people are very comfortable with.
It also has the advantage in lists that when you step through a thread
the answer to the last item is ready for you to read.
I disagree completely, and I am a threaded reader.
The
On Monday 14 June 2004 23:03, twisted wrote:
Friends, Romans, Countrymen, lend me your ears!
This reminds me of the Robin Hood, Men in Tights scene where Robin of Loxley
says the same and all the villagers throw ears at him.
That's disgusting!
And now, for the Asterisk-Users dial plan:
You
Hi Steve,
please could you post your zapata.conf and zaptel.conf files?
Regards
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 15 June 2004 10:28
To: '[EMAIL PROTECTED]'
Subject:
On Tue, 15 Jun 2004, Andrew Kohlsmith wrote:
On Tuesday 15 June 2004 03:17, Holger Schurig wrote:
By the way: what is on channel 0 ?
E1s start at channel 0?
I don't know anything about E1s, but with T1 PRI the D channel is usually the
last channel (channel 24 for PRI). I imagine he's
You are trying to compile it with an out-dated Asterisk
source tree. Use Asterisk CVS HEAD checkout of 2004-06-07.
Michael.
Michael M. Saunders wrote:
Does anyone have any ideas why this is failing
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael M.
I'll give that a try and let you know
Matt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: 15 June 2004 12:21
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Queue then Voicemail
On Tuesday 15 June 2004 05:19, Matt wrote:
On Tue, 2004-06-15 at 12:29, Simon Brown wrote:
And didn't the original poster of this thread state rather forcefully that
this list is for * issues, not to be hijacked - which is exactly what is
happening based on comments/demands made by the original poster that were not
on the topic of *
Certainly, here they are (I've stripped the commented bits away):
Zapata.conf
[trunkgroups]
[channels]
language=en
context=default
switchtype=national
overlapdial=yes
signalling=fxo_ls
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
I am trying to build asterisk and having an odd problem compiling it.
I got the source this morning from CVS with make update. I then did
a make, but I'm getting errors in app/app_voicemail.c...
On line 155 begins a section which almost looks like patch material.
The first line is:
Hello,
I am new in Asterisk... and I didn't get to find this information in the
site or in the links that appears in the support section...
The one that I want to know is which are the compatible softwares with
Asterisk (PC working as an extension of the PBX)...
Theoretically, every H.323
LIFO Last In First Out
- Original Message -
From: tmpm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 15, 2004 4:30 AM
Subject: Re: [Asterisk-Users] Asterisk-Users List Etiquette
Its a matter of personal preference Holger, most people dont care, but the
ones who do
I hate asci signatures that are hard to read, stop eating up my bits :p
- Original Message -
From: Kevin Walsh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 15, 2004 6:07 AM
Subject: RE: [Asterisk-Users] Asterisk-Users List Etiquette
Steven Critchfield [EMAIL
Or compile the .so with -lpq option.
- Original Message -
From:
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 15, 2004 5:06
AM
Subject: Re: [Asterisk-Users] Prepaid
application error
Hi, you have to launch the
script prepaid-make.sh in the
Apollon-Stephan/others--
Thanks-- just what the doctor ordered.
Well get on those solutions today.
Just to complete the thread--
If anyone else has ideas, ie the names of Mac or PC software which produces
the correct GSM file-- PLEASE POST it to this thread.
Also-
If anyone has produced the
On Tue, 15 Jun 2004 08:09:42 -0400, Michael George
[EMAIL PROTECTED] wrote:
I am trying to build asterisk and having an odd problem compiling it. I
got the source this morning from CVS with make update. I then did a
make, but I'm getting errors in app/app_voicemail.c...
On line 155
On line 155 begins a section which almost looks like patch material.
The first line is:
app_voicemail.c
Does
rm app_voicemail.c
cvs co app_voicemail.c
help?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Old managers will change its the LaLawyershat don't
change. Every dam law office that I been in has at
least one fax machine that is constantly printing
something out. But to say fax is dead is an
understatement.
ATT said that about teletype service, you know 50 -
300 baud service, years ago and
Gonzalo,
i would like if some could help me with a * and Mediatrix configuration...
i have this in my extensions.conf file
[outbound]
ignorepat = 9
exten = _901,1,Dial(SIP/[EMAIL PROTECTED])
exten = _901,2,Congestion
exten = _9020,1,Dial(SIP/[EMAIL PROTECTED])
exten =
I'm getting this message when I start Asterisk
chan_capi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81
but when I try and recompile I get this
chan_capi.c:60: error:
`__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__'
undeclared here (not in a function)
any help would
Hi!
It seems that if Asterisk recognises the DTMF digits, it will intercept them
and not send them to the ISDN card (either that, or the ISDN card isn't
regenerating them).
My guess: isdn4linux is the culprit.
Cheers, Philipp
___
Asterisk-Users
Hi!
Now it is pretty obvious that my setup is ok since it work half the
times.
Maybe, maybe when the IPs on dynamic servers change, * has different
information internally hence the transfer fails?
My feeling is that you have a firewall/NAT issue. Look at the qualify=
paramter and do some
I think many of the list members have it, but for their own benefit.
Like jeremy mcnamara from nufone...
HC.
- Original Message -
From: usedcanon [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 14, 2004 5:43 PM
Subject: RE: [Asterisk-Users] SIP prepaid
I am doing something
On Tue, 15 Jun 2004, Michael George wrote:
On line 155 begins a section which almost looks like patch material.
The first line is:
app_voicemail.c
and then there is some C code defining adapp and adsec, then
===
and some more C code defining adapp and adsec again, followed by
Isn't it odd as hell the same people that complain about html are also
some of the same people that use special mail readers to emulate news
readers? Both seem to want to influence the 8,000 list members their
tools are the only one's in existence and we better all format our
list postings to make
Do the Polycom IP phones have some programmability so you can do some
programmable phone buttons like you can on the Cisco phones?
If there is programmability, such as for soft-keys and the like, how
would you rate Polycom's vs Cisco's capabilities? And where can one
find the programming
On Tuesday 15 June 2004 08:29, Kurt wrote:
Old managers will change its the LaLawyershat don't
change. Every dam law office that I been in has at
least one fax machine that is constantly printing
something out. But to say fax is dead is an
understatement.
ATT said that about teletype
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kurt
Old managers will change its the LaLawyershat don't
change. Every dam law office that I been in has at
least one fax machine that is constantly printing
something out. But to
Good Afternoon Everyone,
I am having a problem with compiling the CVS version of *-addons downloaded
today. I am also having problems compiling an older version as well but im
ignoring that one for now.
I believe I have all the correct libraries, and I have done extensive searches
everywhere I
On Jun 15, 2004, at 8:21 AM, Stuart Grimshaw wrote:
On Tue, 15 Jun 2004 08:09:42 -0400, Michael George
[EMAIL PROTECTED] wrote:
I am trying to build asterisk and having an odd problem compiling it.
I got the source this morning from CVS with make update. I then
did a make, but I'm getting
Dave Cotton schrieb:
I'm getting this message when I start Asterisk
chan_capi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81
but when I try and recompile I get this
chan_capi.c:60: error:
`__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__'
undeclared here (not in a
Hi Steve,
SH The presentation is 3 digit, I've made an entry in extensions.conf
SH to allow the looping of a call back from the GDK.
3 Digits seems a bit short from Telewest, I would have expected the last six
digits to have been sent for inbound PSTN calls (as per the BT standard). If
Telewest
; goto philly q
exten = 0,1,Answer
exten = 0,2,Background(wrn-phillyq)
exten = 0,3,Queue,phillyq
exten = 0,4,WaitMusicOnHold(90)
exten = 0,5,Voicemail(u1)
exten = 0,6,Playback(vm-goodbye)
exten = 0,7,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Yesturday I was trying to install the app_prepaid but a guy on irc told me
that its fake and it dosent work. So I start trying to install the
appradius and installed it succesfully, but I cant make it work! Any one
has experience with app_prepaid or appradius that can tell us what are the
Kurt wrote:
Old managers will change its the LaLawyershat don't
change. Every dam law office that I been in has at
least one fax machine that is constantly printing
something out. But to say fax is dead is an
understatement.
ATT said that about teletype service, you know 50 -
300 baud service,
[EMAIL PROTECTED] 6/15/2004 7:29:33 AM
Old managers will change its the LaLawyershat don't
change. Every dam law office that I been in has at
least one fax machine that is constantly printing
something out. But to say fax is dead is an
understatement.
ATT said that about teletype service,
On Tue, 2004-06-15 at 15:02 +0200, Deti Fliegl wrote:
Please apply the patch attached to this mail.
Thanks that got it compiled, but I still see this error
chan_capi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81
any thoughts?
--
Dave Cotton [EMAIL PROTECTED]
Polycom IP 600's are fully programmable, much more so than the Cisco
phones. Yes, you can program the phone buttons. That and just about
everything else you can imagine is programmable via xml configuration files.
Goto http://www.polycom.com/common/pw_item_show_doc/0,1276,2545,00.pdf
for the
No, the presentation is 3 digits currently from Telewest.
We only have 15 channels coming from Telewest (the link back from the GDK to
Asterisk could use all 30, but I don't think we ever get close to the 15 so
I just configured it the same on both ends for now).
I'll make those changes this
On Tue, 15 Jun 2004 09:07:19 -0400, Michael George
[EMAIL PROTECTED] wrote:
Okay, that was it. Just my inexperience with CVS shining through :)
Thanks for the help!
Now I just have to remember why I changed it and if I need to change it
again...
you could always do cvs diff filename and
On Tue, Jun 15, 2004 at 07:35:31AM -0600, Rich Adamson said:
Isn't it odd as hell the same people that complain about html are also
some of the same people that use special mail readers to emulate news
readers? Both seem to want to influence the 8,000 list members their
tools are the only
I grabbed the lastest CVS and it stilled failed.
Would you be able to give me the command to get 2004-06-07
Because when I login I can only get it by release numbers.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Manousos
Sent: Tuesday, 15 June
Polycoms do have programmable softkeys as well as several other programmable
features. They don't however have expandable button modules like the Ciscos
do to add extra physical buttons to push.
Their big strengths are that they are wonderful for conference rooms and for
use as a speakerphone and
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 15 June 2004 14:59, Luckcuck Nick-LCKN001 wrote:
CFLAGS+=-fPIC
CFLAGS+=-I../asterisk
CFLAGS+=-I/usr/include
Don't know if it helps, but try adding:
CFLAGS+=-I../asterisk/include
- --
Regards,
Tais M. Hansen
ComX Networks
Tel:
please update to 0.3.4a.
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/
Am Mo,
I'm having problems getting Asterisk SIP to register with an Entice
softswitch SIP Gateway. My provider tells me that all thats needed is a
user name, password and the IP address and to register and it needs to
be using MD5 authentication.
I continualy get a 603 Decline message. The provider of
At 15:59 15/06/2004 +0200, you wrote:
On Tue, 2004-06-15 at 15:02 +0200, Deti Fliegl wrote:
Please apply the patch attached to this mail.
Thanks that got it compiled, but I still see this error
chan_capi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81
any thoughts?
You could do a capi
On Jun 14, 2004, at 4:10 PM, Michael George wrote:
Has anyone had any luck with this application? I've tried it under
both pass and fail conditions and it just hung asterisk on me...
I've worked with this a bit more and if the mailbox does exist, it will
go to priority+101 as it should.
Hi I am trying to use SetCDRUserField in an agi script
but with no success.
I am using the CDR mysql addon, however I can't see it
being at fault as my attempt is not doing anything to
the CVS CD either.
has anyone used this, any hints guidence would be
greatly appreciated.
The syntax I am
I got it working (but not realy)...
I can authenticate with a Prepaid Card Number (10 digits) and it
tells(see asterisk debug -cvvv,-rvvv) me to enter destination number.
But i'm lazy to fill allowed destinations database but i think that it
will work too.
Alex
On Tue, 2004-06-15 at 15:35
cvs co -D 2004-06-07 asterisk
Michael M. Saunders wrote:
I grabbed the lastest CVS and it stilled failed.
Would you be able to give me the command to get 2004-06-07
Because when I login I can only get it by release numbers.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On Tue, 2004-06-15 at 08:35, Rich Adamson wrote:
Isn't it odd as hell the same people that complain about html are also
some of the same people that use special mail readers to emulate news
readers? Both seem to want to influence the 8,000 list members their
tools are the only one's in
Tais,
Genius.. I honestly thought I had tried every combination of includes :P
Cheers.
--
Nick
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tais M. Hansen
Sent: 15 June 2004 15:24
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cdr_addon_mysql.c
On Tue, 2004-06-15 at 15:26 +0100, Jason Williams wrote:
You could do a capi debug in the CLI and see if you can get more
information on the error.
This error is showing up on loading the capi module.
Dave
___
Asterisk-Users mailing list
[EMAIL
Hi I am trying to use SetCDRUserField in an agi script
but with no success.
I am using the CDR mysql addon, however I can't see it
being at fault as my attempt is not doing anything to
the CVS CD either.
I don't think that SetCDRUserField works on MySQL :( I only implemented
it for CSV.
On Tue, 2004-06-15 at 11:43, Shaun Ewing wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jason Williams
Sent: Tuesday, 15 June 2004 6:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Outgoing DTMF when using BRI
i4l
On Jun 15, 2004, at 7:27 AM, mattf wrote:
Polycoms do have programmable softkeys as well as several other
programmable
features. They don't however have expandable button modules like the
Ciscos
do to add extra physical buttons to push.
While Cisco does make an expansion module for the 7960, it
Hi I am trying to use SetCDRUserField in an agi script
but with no success.
I am using the CDR mysql addon, however I can't see it
being at fault as my attempt is not doing anything to
the CVS CD either.
I don't think that SetCDRUserField works on MySQL :( I only implemented
it for
Does it do the cost calculation by destination (country rates)? if yes it
should be great! Does it work with sip clients to or only with digium
cards! also if it is posible to tell us what should we do to make it work,
should we add some lines to extensions.conf?
Best Regards
Hekuran Doli
I
Hello,
the new Soundpoint 1.2 SIP release is on the Asterisk Polycom site for
download:
http://www.freedomphones.net/polycom/files/SoundPoint-IP_SIP_1.2.0.zip
included with it is the new Admin Guide.
Other older Polycom Soundpoint files are also available for download on the
site:
On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote:
Ive done something similar at home, but made my dialplan such that I
can dial either 10 or 11 digits locally. I dont use a throw away
digit at all. Any 7, 10, or 11 digit call will be appropriately
mangled and sent out the PSTN / VoIP
I installed app_prepaid and it works,
but it is for calling card bussines (in-band) authentication, so you dial,it
answers and ask you for a PIN to dial.
What I wasn looking for is a prepaid app for SIP users.
I tryied to modify prepaid_app but it is based on postgreSQL, and I have
mysql.
HC.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Robert Withrow
Sent: Tuesday, June 15, 2004 12:32 PM
To: Asterisk-users
Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata-
186)
On Mon, 2004-06-14 at 19:34,
After over a month (well, ok, no more than an hour a day :) of planning,
getting hardware, tinkering and testing, I'm about to my Ultimate Home
Phone System (tm) online.
Connectivity to the outside world is provided by:
A. 1 POTS phone line connected through an X100P ($11/month, needed to
carry
I use DISA on the asterisk box and have the dialplan on the ata set
so that calls starting with 9 or 8 have only two digits.
disa extensions 90 - 99 are for pstn calls via various providers.
Those in 80 - 89 are for fwd and other similar services.
The ata's dialplan looks like:
DialPlan:
The issue we have here is not just related to IAX. If you have Asterisk
step out of the media stream for any call, you lose the capability to
determine the status of the call, and therefore lose the ability to track
the call in your CDR.
Perhaps (at least for the case of IAX transfers or
On Tue, 2004-06-15 at 12:58, Jay Milk wrote:
After over a month (well, ok, no more than an hour a day :) of planning,
getting hardware, tinkering and testing, I'm about to my Ultimate Home
Phone System (tm) online.
However, my concern is that Asterisk will keep its Zap channels straight
-- I
They will keep the same configuration. It's the magic tech faeries that
make it work as such. :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Tuesday, June 15, 2004 1:58 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Multiple X100Ps
On Tuesday 15 June 2004 13:53, Chris A. Icide wrote:
The issue we have here is not just related to IAX. If you have Asterisk
step out of the media stream for any call, you lose the capability to
determine the status of the call, and therefore lose the ability to track
the call in your CDR.
Chris A. Icide wrote:
The issue we have here is not just related to IAX. If you have Asterisk
step out of the media stream for any call, you lose the capability to
determine the status of the call, and therefore lose the ability to
track the call in your CDR.
That brings up a question that's
Jon Radon wrote:
They will keep the same configuration. It's the magic tech faeries that
make it work as such. :)
They will keep the same order _most_ of the time, however there are
things out of your control that can cause the order to change:
recompiling your kernel to use ACPI when you did
I've searched the lists but I didn't find anything exactly like this.
I have two Grandstream BT101 phones connected to an Asterisk.
Periodically, for reasons that I can't determine, one or the other (or
both) of the BT101s decide(s) to go on permanent busy. Dialing that
phone gives:
--
1 - 100 of 151 matches
Mail list logo
