Hey all,
I was wondering if anyone knows about a Mexican ITSP I can connect to to
route calls from and to my * boxen.
If it matters: I'm located in The Netherlands and one of our customers
is in Mexico so if we need a Mexican presence that is not an issue.
Thanks.
--
Michiel van Baak
Why are you putting semi-colons at the end of every line? The dialplan
isn't written in PHP ;).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L.
Kline
Sent: 15 July 2009 23:46
To:
I have set my Asterisk server up to connect to my Jabber server and
send messages with the caller ID details in them to the recipients of
incoming calls - this is working very nicely.
There are a few other things I can think of right now that I would
like to send to Jabber but as yet I do
Jerry Geis wrote:
The PBX guy seems to always complain about how he has MANY options
and thats not enough information...
What else am I supposed to supply this person. Are they not the PBX
expert?...
Anyway as example. the last customer I told the above information. He
set up the PBX
and
I would like to have the ability to have Asterisk announce the temperature
-- not using TTS -- within the dialplan.
For a non-Asterisk project, I have a cron job that periodically pulls down
an XML file from weather.com containing local weather data (TWC's user
agreement requires that data be
Hi all,
I'm sure this has been done before but I just can't figure it out.
On my * box I have a simple IVR:
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten =
Sebastian Maz wrote:
this is what I'm trying to accomplish:
- receiving an inbound call from A
- dialing another number (B)
- bridge A and B
- every x minutes, debridge A and B, and bridge A with C (SIP call to
an platform that is gonna play an ad)
- rebridge A and B
Any advice
John A. Sullivan III wrote:
On Wed, 2009-07-15 at 14:34 +1000, Alex Samad wrote:
The subject line says it all how do I enable this style of call.
Pointers to the dns setup and asterisk setup would be great
snip
If I understand what you are seeking, you can try these URIs:
I have just the thing in PHP.
Drop me a personal e-mail and I'll whiz it over.
Andrew Thomas
Technical Services Manager
a...@datavox.co.uk
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF
-Original Message-
From:
Check this one out.
developed for AstLinux, it ought to be close to what you want.
depending on your version, you may need to modify sound file references
http://lonnie.abelbeck.com/astlinux/info/weather.php
John Novack
Trevor Hammonds wrote:
I would like to have the ability to have Asterisk
Each of these should be do-able either through dialplan snippets, cron jobs
or AGI's. (a) would be a dialplan snippet (b) would be a dialplan snippet
(c) would require an AGI or cron to monitor how long the peer has been out
of service.
-Original Message-
From:
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Hash: SHA1
Trevor Hammonds wrote:
I am hoping someone on the list has an example of a lightweight AGI script
that I may modify to either read the simple text file and set a dialplan
variable to the current temperature, or hopefully a more-sophisticated one
I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax
(.tiff) from the first
asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an
INVITE with audio G.711. Asterisk2
accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to
On Wed, Jul 15, 2009 at 5:23 PM, Waynewa...@planetwayne.com wrote:
Hi all,
Just a quickie to say that this has been solved now - real simple -
downloaded '*current*' rather than the versions from the home page of
Astrisk.org. (didn't realise there was a 'current' version tbh.
Anyways - I
Barry L. Kline wrote:
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Mark Michelson wrote:
You need to set a call-limit for the SIP peer. Device state calculation for
a
SIP peer is predicated on both the call-limit and busylevel. Let's say that
you
were to have a call-limit of 2, but
hutx wrote:
I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax
(.tiff) from the first
asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates
an INVITE with audio G.711. Asterisk2
accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with
Hi all,
I have this installation:
Asterisk 1.6.1.1 with h323 support, pwlib_v1_10_3 and openh323_v1_18_0.
I have a problem that is, when a call comes from H323 and goes to a Sip
phone the asterisk sends two rtp streams to the sip. I checked this with
tcpdump, save the payload (voice is in
Hi all,
I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled
voicemail in the extensions area, and set the default password. However,
every time I try to log in with a mailbox and password, I get the message
login incorrect. I've tried changing the voicemail password, and
Hi Zeeshan,
You might want to take a look at our solution here: http://www.flashsip.com/
We do the customization of the software for our clients on demande.
Best regards,
Brian
On Sat, Jul 11, 2009 at 2:55 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
For a while now I've been looking for a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mark Michelson wrote:
Thanks for the config info. I have a couple of suggestions for fixes.
1. Try changing the type in [basic-options] from friend to peer. I've found
that
device state reporting for outbound calls (from the perspective of
Hi, I've managed to get HYLAFAXT38MODEM-ASTERISKCISCOAS5400
working, but when they are negotiating asterisk drops a message telling
Unknown RTP codec 96 received from gateway Do somebody know how to fix it
?
Thank you !
[ TYPE: Control (4) SUBCLASS: Ringing (3) ]
On Thu, 2009-07-16 at 10:57 -0400, Zaheer Master wrote:
Hi all,
I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled
voicemail in the extensions area, and set the default password. However,
every time I try to log in with a mailbox and password, I get the message
login
Thanks for the reply John. In the voicemail.conf file there were two extra
[] creating a NULL context. Removing those extra brackets fixed the problem.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
Xavier Cardil wrote:
Hi, I've managed to get HYLAFAXT38MODEM-
ASTERISKCISCOAS5400 working, but when they are negotiating asterisk
drops a message telling Unknown RTP codec 96 received from gateway Do
somebody know how to fix it ?
There's nothing to fix; the gateway sent an
Huh?
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz
is not the same as
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.10.tar.gz?
Their sha1 files are identical.
sean
I believe he means that:
Try http://www.inext.com.mx they can provide DIDs in several cities in
Mexico.
On Thu, 2009-07-16 at 09:16 +0200, Michiel van Baak wrote:
Hey all,
I was wondering if anyone knows about a Mexican ITSP I can connect to to
route calls from and to my * boxen.
If it matters: I'm
Xavier Cardil escribió:
Hi, I've managed to get HYLAFAXT38MODEM-
ASTERISKCISCOAS5400 working, but when they are negotiating
asterisk drops a message telling Unknown RTP codec 96 received from
gateway Do somebody know how to fix it ?
Thank you !
[ TYPE: Control (4)
Hi Kelvin, thank you for your response, well in fact it is not working but
that's only a NOTICE, not an error. Warnings comes after that and the fax is
not sent. Take a look at the last lines of this output :
[ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-01ba5638]
IVR3*CLI debug
Michael Graves wrote:
On Tue, 14 Jul 2009 22:58:14 + (UTC), Jeff LaCoursiere wrote:
Jeff, yeah i saw the posts, i followed Bob Pierce config and had no
luck, BUT it just started to work, i changed AP's, seems like theres
something wrong with Ubiquiti NanoStation2 WMM implementation, i
Thank you for the heads up. I will look into both weephone and voipover3g
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi,
I would like to know if someone can suggest me an efficient way of writing a
dialplan to match variable-length international calls when using SIP clients
with the early dial or 484 feature.
What I usually do for clients that do NOT early dial is define something like
this in my outbound
The documentation in
http://svn.digium.com/svn/asterisk/branches/1.4/configs/iax.conf.sample
(and http://svn.digium.com/svn/asterisk/branches/1.6.*/configs/iax.conf.sample)
seems slightly wrong.
---
; ... Limited IP based
; access control is allowed by use of allow and deny keywords. ...
On Thursday 16 July 2009 12:26:25 Philipp Kempgen wrote:
I think this does not justify filing a bug.
No, it does. Go ahead and file it.
--
Tilghman Teryl
with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies)
and Harry, BB, George (dogs)
At 06:37 AM 7/15/2009, you wrote:
Ours is just internal, but the concept should be the same. My boss could
talk on his phone fine until he cranked up Foxnews feed. Once the video
started, he couldn't talk on his phone anymore (bad quality or total loss of
call).
What I've done here is probably
On Thu, 2009-07-16 at 04:49 -0700, Trevor Hammonds wrote:
I would like to have the ability to have Asterisk announce the temperature
-- not using TTS -- within the dialplan.
Chapter 9 of Asterisk: The Future of Telephony shows you how to build
an AGI script to do just that. For a free
Tilghman Lesher schrieb:
On Thursday 16 July 2009 12:26:25 Philipp Kempgen wrote:
I think this does not justify filing a bug.
No, it does. Go ahead and file it.
ok. https://issues.asterisk.org/view.php?id=15518
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -
I managed to do it with a simple forward to the one of our DIDs The
problem now is that I loose the CID of the original caller ... Is there a
way to forward the call with the original caller ID ?
Thanks!
G.
On Wed, Jul 15, 2009 at 5:38 PM, John Novack
jnov...@stromberg-carlson.orgwrote:
Philipp Kempgen philipp.kemp...@amooma.de writes:
Benny Amorsen schrieb:
Last concern: Does setvar work even for transfers, like accountcode
does?
I can't answer your question, but transfer != transfer. Some use
a feature code in Asterisk, some initiate a transfer on their phone,
some use
Looks like the caller ID gets lost when you forward. Normally if you do a
simple forward using central office features, the caller ID will be the
calling party's number. If you're using a PBX (or something that looks like
one) the PBX does a hook-flash, makes a call to your PRI DID and you see the
Hello everybody!
Please let me ask you a question:
Is it possible (and if yes, how) to configure 2 asterisk servers on two
machines so that the second one acts as a backup system if the first one
is unresponsive?
Clearly, the second should take over automacigally (but not necessarily
during an
- Norbert Zawodsky norb...@zawodsky.at wrote:
Hello everybody!
Please let me ask you a question:
Is it possible (and if yes, how) to configure 2 asterisk servers on
two
machines so that the second one acts as a backup system if the first
one
is unresponsive?
Clearly, the second
I'm going to give a qualified no. The reason being is that setvar works
in a session (say SIP/100-abcdefg) and the blind transfer may spawn a new
session like Local/1-abcdefg). So your only solid variables are the global
ones. You can verify this by looking at CLI output with verbose set to at
On Tue, 2009-07-14 at 00:01 +0200, Benny Amorsen wrote:
Last concern: Does setvar work even for transfers, like accountcode
does?
At least in theory, the setvar= setting in sip.conf or iax.conf (or in
Asterisk 1.6.0 and later, chan_dahdi.conf) should work just like the
Set() dialplan
Apologies for the off topic post... hoping someone knows if 800 number
portability in the states is legally enforced? One of my customers is
being told by their current vanity 800 provider that they own the number
and refuse to release it to their new carrier. I thought I understood
that in
Hello,
I have an application that needs to record outgoing calls. It's
running on Asterisk 1.4.18, with CDR data stored in MySQL.
Outgoing calls are recorded based on their uniqueid. When outgoing
calls are placed, there is a line like this on my extensions.conf:
exten =
Hello,
We have an application where operators will sometimes take an incoming
call from a queue, then contact an outside line, do a consultation,
and finally do a SIP attended transfer to join the two parties
together. We'd like to record the incoming caller's conversation with
the operator and
I don't know the full details, but I think if the Dial command(s) have the W
and/or w options on them, you can activate/deactivate recording via DTMF.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Scott
On Thu, Jul 16, 2009 at 5:39 PM, Jeff LaCoursiere j...@jeff.net wrote:
Apologies for the off topic post... hoping someone knows if 800 number
portability in the states is legally enforced? One of my customers is
being told by their current vanity 800 provider that they own the number
and
Changing toll-free RespOrgs (Responsible Organizations) is different from
number portability.
That said, the owner of a toll-free number has the right to change RespOrgs,
so the question is Who is the owner?
Has your customer been buying simple toll-free service and owned the number
all along,
Try to get a level one tech to set RDNIS on your forwarded POTS line.
Good luck!
On Thu, Jul 16, 2009 at 3:30 PM, Don Kelly d...@donkelly.biz wrote:
Looks like the caller ID gets lost when you forward. Normally if you do a
“simple forward” using central office features, the caller ID will be
Hi all
I'm trying to install asteris 1.4.22.1 on Solaris 10...the server is V120
SUN spark...During compilation (gmake) I got the following error
/vis.c -o np/vis.o_a
np/vis.c: In function `svis':
np/vis.c:205: error: `u_int32_t' undeclared (first use in this function)
np/vis.c:205: error: (Each
On Thu, 16 Jul 2009, Don Kelly wrote:
Changing toll-free RespOrgs (Responsible Organizations) is different from
number portability.
That said, the owner of a toll-free number has the right to change RespOrgs,
so the question is Who is the owner?
The owner in this case is CallSource
There are national number rental agencies that lease out prime 800 numbers
even down to the rate center level.
They own the number, not the renter, and there is a contract that says so.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Danny Nicholas da...@debsinc.com writes:
I don't know the full details, but I think if the Dial command(s) have the W
and/or w options on them, you can activate/deactivate recording via DTMF.
Thanks, that's a good idea, I might be able to rig something up with
that!
Scott.
I've been using 1.6.2 for a few weeks and I've managed to get almost
everything working perfectly.
I can't get the MWI indicators on my Aastra phones to work properly,
the did in all the versions of 1.2 I used up to the most recent one,
but now they work correctly right after the phone is
In the good old days telcos didn't care how many channels your forward
used up, they just did it. However nowadays they only allow one
channel at a time to be forwarded, if you need more you have to pay
for it.
Verizon here in NJ charges around $8.00 a month for each call path
(channel), and so do
On Thursday 16 July 2009 20:19:07 Ira wrote:
I've been using 1.6.2 for a few weeks and I've managed to get almost
everything working perfectly.
I can't get the MWI indicators on my Aastra phones to work properly,
the did in all the versions of 1.2 I used up to the most recent one,
but now
On Thu, 16 Jul 2009, Don Kelly wrote:
Changing toll-free RespOrgs (Responsible Organizations) is different from
number portability.
That said, the owner of a toll-free number has the right to change
RespOrgs,
so the question is Who is the owner?
The owner in this case is CallSource
On 11:39, Thu 16 Jul 09, Carlos Chavez wrote:
Try http://www.inext.com.mx they can provide DIDs in several cities in
Mexico.
Thanks.
I asked the customer to have a look (I'm only capable of reading English
and Dutch ;))
You have any experience with them ?
On Thu, 2009-07-16 at 09:16
C F wrote:
If you don't want to port it to the PRI for whatever reason you can
convert it to a RCFW (remote call forwarded number) which is around
$15.00 plus $8.00 for each additional channel again pricing is for
here in Verizon land.
Is that true even if the number is out of a rate center
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