Greetings all-
I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks
rather interesting. Has anyone used one? Where did you purchase it? Pricing?
Operational issues?
http://spidermux.com/
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
--
Philip A. Prindeville wrote:
Here's a segment of my dialplan, I'm working on the freenum/ISN
functionality:
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})
same = n,GotoIf($[${isnresult} != ]?:fn-CONGESTION,1)
; set up our outgoing call state
same =
Hi,
I am new to asterisk and trying to make calls with TDM400P asterisk digium
card.
I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and
libpri-1.4.10.2 packages which are downloaded from asterisk website (
www.asterisk.org)
and able to compile successfully. TDM400P Digium
Hello, i have this problem :
i phone person B .
*if i hang up*, i have this h extension : exten = h,1,AGI(ende.agi)
*if the person B hangs up* , i have this h extension : exten = h,1,
DeadAGI(ende.agi)
The problem is, i do not know where hangs up the first . How kann i combine
AGI and DeadAGI in
Hello, i have this problem :
i phone person B .
*if i hang up,* i have this h extension : exten = h,1,AGI(ende.agi)
*if the person B hangs up* , i have this h extension : exten = h,1,
DeadAGI(ende.agi)
The problem is, i do not know where hangs up the first . How kann i combine
AGI and DeadAGI in
Thanks Loan Indreias ... Nice Idea
Thanks Danny Nicholas.
Cheers
On Tue, Apr 27, 2010 at 7:17 PM, Danny Nicholas da...@debsinc.com wrote:
This is probably a good idea, BUT it is likely that the dialed phone will
never ring (Perhaps that is the desired effect); In my experience it takes
Redouane Zerargui wrote:
Hello, i have this problem :
i phone person B .
_/*if i hang up*/_, i have this h extension : exten = h,1,AGI(ende.agi)
_/*if the person B hangs up*/_ , i have this h extension : exten =
h,1,DeadAGI(ende.agi)
The problem is, i do not know where hangs up the
Hello. Where i can find complete realtime mysql schema for asterisk 1.6?
Google get results to some tables.
I want to do all
iaxusers
iaxpeers
sipusers
sippeers
sipregs
voicemail
extensions
meetme
queues
queue_members
musiconhold
queue_log
in separate mysql tables.
--
Vasiliy G Tolstov
I have got call recording working on our 1.4.30 asterisk box together
with a recording pause ability and being able to play different audio to
each party at the start and end of the pause. This all works perfectly
but one wish is to have the audio files have a beep or something in them
so when
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 29, 2010, at 1:33 AM, Ishfaq Malik wrote:
Philip
Ignore me I figured it out. The dangers of copy and paste.
After looking through the code line by line I noticed the 'b' parameter
to monitor(). Fine to use before the dial command but shouldnt be used
when a call is in progress.
Gareth Blades wrote:
I have got call recording working on our
I was just wondering if anyone is having the same problem will Polycom 330 ip
phone. The phone looses the network and when you reboot the phone it can no
longer find the DHCP server. I put an address in manually, but the phone is
still not able to connect to the network. I replaced the phone
Aditya Kumar wrote:
re-posting the question.
---
use case:
when some one in my pbx calls 100.200, I have translations well defined,
Media also (media via asterisk) --Works.
when some one calls bob, or for any names I am adding Domain and call is
been sent to the other party --
Hi,
I need a feature from asterisk with dahdi channels, if there is an
incoming call, it should ring on several dahdi channels.
My channels look like:
OFFICE1=DAHDI/13,,rtT
OFFICE2=DAHDI/14,,rtT
If I add this line:
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2})
only OFFICE1 rings.
If I
2010.04.20. 16:50 keltezéssel, Shaun Ruffell írta:
On 04/19/2010 03:48 AM, Peter Gelencser wrote:
I've run into a veird problem. I'm using a B400P BRI and an A1200P card
with dahdi (2.2.1) driver. The dahdi_scan shows the each moduls and
spans, everything seems fine. With dahdi_genconf I
Try this.
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2},,rtT)
Peter Gelencser wrote:
Hi,
I need a feature from asterisk with dahdi channels, if there is an
incoming call, it should ring on several dahdi channels.
My channels look like:
typo ...
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}${OFFICE2},,rtT)
Gareth Blades wrote:
Try this.
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2},,rtT)
Peter Gelencser wrote:
Hi,
I need a feature from asterisk with dahdi
Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
Right! Be very careful about what you're matching! When it comes to matching
things like 'N', 'X', 'Z',
Good snippet, Leif. It's easier to read 100 threads on this forum than the
100 pages of the infamous Asterisk Book PDF.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Thursday, April 29,
Possibly or possibly not. Most (IMO) calls are placed initially with the
choice 2-3 or more codecs. Normally one codec is negotiated and life goes
on, but IAX is a little different from a SIP/DAHDI call. The most certain
remedy I can think of for this it to just unallow the alaw codec on IAX
Hi,
What does this message imply?
[Apr 29 14:46:30] NOTICE[32175] channel.c: Dropping incompatible voice frame on
IAX2/trunk1-9085 of format alaw since our native format has changed to 0x4
(ulaw)
If voice frames have been dropped then I suppose that the call quality may be
affected?
Vieri
Greetings List.
I'm facing a strange issue with one of my providers.. after sending an INVITE
request my server places the call on hold.. until the call is answered..
this is happening only with this provide although i have 3 other providers i
route calls through..
can anyone explain what is
Can you post a sip debug
Tarek Sawah wrote:
Greetings List.
I'm facing a strange issue with one of my providers.. after sending an INVITE
request my server places the call on hold.. until the call is answered..
this is happening only with this provide although i have 3 other providers i
On 4/29/10 4:22 AM, Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
That's capital N, isn't it?
Also, the prefix _stdexten-. seems to
It's a pattern matching thing; the asterisk module knows how to process
stdexten, but thinks that n or N is a digit substitution. When n or
N is escaped ([n] or [N]) the program knows to treat it as a literal and
not a pattern match.
-Original Message-
From:
The phone is only making one call, notice the call-id did not change.
The second INVITE is sent in responce to a 401 Authentication
Required. The 401 will contain the necessary authentication
information for the phone to use to build the Authorization header
that it inserts in the second invite.
On Wed, Apr 28, 2010 at 09:34:18AM -0700, Steve Edwards wrote:
On Wed, 28 Apr 2010, Ryan Bullock wrote:
Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early
in the script to read in anything from stdin?
(From the docs)
# pull AGI variables into %input
%input =
On Wed, 28 Apr 2010, Ryan Bullock wrote:
Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early
in the script to read in anything from stdin?
(From the docs)
# pull AGI variables into %input
%input = $AGI-ReadParse();
On Wed, Apr 28, 2010 at 09:34:18AM -0700, Steve
Speaking from a Perl'er perspective, there's no good reason that
Asterisk::AGI shouldn't do the ReadParse automatically except that it
requires the module author to do something that the user should be doing as
a best practice and could lead to unexpected errors in a reuse
environment. IMO there
On Thursday 29 April 2010 11:46:39 Philip Prindeville wrote:
On 4/29/10 4:22 AM, Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
That's capital N,
On 04/29/2010 12:09 PM, Tilghman Lesher wrote:
On Thursday 29 April 2010 11:46:39 Philip Prindeville wrote:
On 4/29/10 4:22 AM, Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean
Un-top-posting...
On Wed, 28 Apr 2010, Ryan Bullock wrote:
Looking at the Asterisk::AGI docs, maybe try calling ReadParse()
early in the script to read in anything from stdin?
(From the docs)
# pull AGI variables into %input
%input = $AGI-ReadParse();
On Wed, Apr 28, 2010 at 09:34:18AM
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in
If you dial 8029 from PBX1, does VM work? In my experience, cross-version
IAX is tricky.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Thursday, April 29, 2010 2:03 PM
To: Asterisk Users Mailing List
In PBX1, where are you actually dialing the phone? The first line of the
macro just says goto dialstatus with no Dial statement.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Thursday, April 29, 2010 2:03 PM
2010/4/29 garge rama garge.r...@gmail.com
Hi,
I am new to asterisk and trying to make calls with TDM400P asterisk digium
card.
I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and
libpri-1.4.10.2 packages which are downloaded from asterisk website (
www.asterisk.org)
I'm considering a situation where I buy about twenty ATA devices.
I've played with the Linksys / Cisco PAP2T, and got that working fine
with some inbound and outbound faxing. The web GUI was okay. I'm
seeing prices around $45 to $50 for this thing. It comes with two FXS
ports, but I only need one
Hi Guys,
Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
Peder: i just didn't want to put a lot of lines, (by the way it's dialing
talking fine), but here you are:
[macro-stdexten]
exten = s,n,Dial(SIP/${ARG1}IAX2/${ar...@${arg1},20,tTrWw);Ring phone
for 20 seconds
exten =
On Thursday 29 April 2010 13:11:17 Philip A. Prindeville wrote:
Doesn't quite make it 'deterministic' if you have to test it to see what
it's going to do.
The code is deterministic. The human who wrote the example is not. Are
you proposing a genetic modification to make humans deterministic?
Worse things have been proposed for humans; many readers would like to see
this done to posters such as I. :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Thursday, April 29, 2010 2:55
On 4/29/10 1:55 PM, Tilghman Lesher wrote:
On Thursday 29 April 2010 13:11:17 Philip A. Prindeville wrote:
Doesn't quite make it 'deterministic' if you have to test it to see what
it's going to do.
The code is deterministic. The human who wrote the example is not. Are
you
On Thu, 29 Apr 2010, David Backeberg wrote:
I'm considering a situation where I buy about twenty ATA devices.
I've played with the Linksys / Cisco PAP2T, and got that working fine
with some inbound and outbound faxing. The web GUI was okay. I'm
seeing prices around $45 to $50 for this
On Thu, 29 Apr 2010, David Backeberg wrote:
I'm considering a situation where I buy about twenty ATA devices.
I've played with the Linksys / Cisco PAP2T, and got that working fine
with some inbound and outbound faxing. The web GUI was okay. I'm
seeing prices around $45 to $50 for this thing.
On 4/29/2010 3:10 PM, Jeff LaCoursiere wrote:
On Thu, 29 Apr 2010, David Backeberg wrote:
I'm considering a situation where I buy about twenty ATA devices.
I've played with the Linksys / Cisco PAP2T, and got that working fine
with some inbound and outbound faxing. The web GUI was okay.
Dan Journo wrote:
On Thu, 29 Apr 2010, David Backeberg wrote:
I'm considering a situation where I buy about twenty ATA devices.
I've played with the Linksys / Cisco PAP2T, and got that working fine
with some inbound and outbound faxing. The web GUI was okay. I'm
seeing prices around
On Thursday, April 29, 2010, David Backeberg wrote:
What do people think about both products?
Bonus points for if people have bulk deployed these, either with TFTP
and configs pushed from a server, or some other good idea.
I can't claim the bonus points. However, I did have a couple of
Hi,
I'm having a major problem with random calls dropping. After spending weeks
trying to figure it out, i've finally spotted the issue but don't know how to
resolve it.
I run a sip server that's hosted in a data centre. It has a public IP address
with no nat involved. My provider also has a
Hi,
Few days ago, my asterisk began to stop unexpectedly
What I did:
- Added a mp3 to the musiconhold directory
- Adjusted the permissions (chown asterisk:asterisk + chmod 755)
- Reconfigured the musiconhold.conf to the deprecated format (found the
example on the internet)
Danny Nicholas wrote:
Good snippet, Leif. It's easier to read 100 threads on this forum than the
100 pages of the infamous Asterisk Book PDF.
Infamous? Ouch :)
Leif.
--
_
-- Bandwidth and Colocation Provided by
On 04/29/2010 03:56 PM, Leif Madsen wrote:
Danny Nicholas wrote:
Good snippet, Leif. It's easier to read 100 threads on this forum than the
100 pages of the infamous Asterisk Book PDF.
Infamous? Ouch :)
Leif.
Danny:
Well, there is an effort to improve the documentation. See
Bryan Jacobs wrote:
I can't just call the car - the car is my cell phone DID with a
bluetooth kit.
I did this same thing you're attempting. I have a desk set at home, a
Polycom in my office and my cell phone all being called at the same
time. I called Verizon and had them disable voice mail
Not really complaining, but AKAIK, this document is current as of about
1.4.10?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philip A.
Prindeville
Sent: Thursday, April 29, 2010 4:59 PM
To:
On 29 Apr 2010, at 22:56, Leif Madsen wrote:
Danny Nicholas wrote:
Good snippet, Leif. It's easier to read 100 threads on this forum than the
100 pages of the infamous Asterisk Book PDF.
Infamous? Ouch :)
He's insulting our holy book! Stone him!
;)
S
--
Certainly getting people to email in updated examples would speed the
book along...
On 04/29/2010 04:06 PM, Danny Nicholas wrote:
Not really complaining, but AKAIK, this document is current as of about
1.4.10?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Are you guys talking about the Asterisk Cookbook Because that
could be released in the next 20 years at this point...
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux
On Thu, 29 Apr 2010 18:03:26 -0400
Barry L. Kline blkl...@attglobal.net wrote:
Bryan Jacobs wrote:
I can't just call the car - the car is my cell phone DID with a
bluetooth kit.
I did this same thing you're attempting. I have a desk set at home, a
Polycom in my office and my cell
On Wed, Apr 28, 2010 at 6:57 PM, James Lamanna jlama...@gmail.com wrote:
Hi,
I have a duplicated DTMF issue with, it appears, bridged IAX channels.
I have the following setup:
PRI IAX
* PSTN ---* Dialplan
I've configured a number on the dialplan server to
Greetings,
I'm trying to continue to do some processing after a TIMEOUT
(absolute). In my dialplan below, when a call comes in to [default],
I call macro-phonenum and pass it a timeout of 20 seconds. macro-
phonenum sets TIMEOUT(absolute), then loops saying the phone number
that was
Hi,
If I was going to post this as an iPhone-only SIP client, I'd expect
loud booing and hissing, but Media5 mobile SIP client is available for
the Symbian S60 platform, too, or will be shortly. Interested? To join
us and hear about Media5 form Pascal Dore, see http://vuc.me
Speaking of mobile,
Hi all...
I have a problem with caller id on my asterisk server.
here is my configuration :
centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2
ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting)
everything fine until I try to feed my app with caller
60 matches
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