On Wed, Apr 11, 2007 at 08:09:16PM -0700, Yuan LIU wrote:
From: Sanjay Rajdev [EMAIL PROTECTED]
Date: Thu, 12 Apr 2007 01:29:51 +0530 (IST)
[good stuff sniffed]
and downloaded zaptel 1.4.1, after that executed the following commands
./configure
make clean
make
make install
Went to
I'm a first time user of Asterisk and have a working setup which I find
clumsy. How can I clean things up to make the dialplan easier to maintain?
My problem
==
I have 6 public numbers that can reach 6 individual users. I have 6
lines like this in sip.conf:
[general]
register =
There are a few mentions in the wiki [1] about a zapata.conf flag
hanguponpolarityswitch. It is meant to cause Asterisk to detect a
hangup when the line polarity switches at the end of the call.
The wiki mentions using the flag in zapata.conf but when I do Asterisk
ignores it:
Apr 12
Nick Adams wrote:
There are a few mentions in the wiki [1] about a zapata.conf flag
hanguponpolarityswitch. It is meant to cause Asterisk to detect a
hangup when the line polarity switches at the end of the call.
The wiki mentions using the flag in zapata.conf but when I do Asterisk
ignores
Hello
Im trying to install an old version of Asterisk.
But it isnt working:
when I run make install:
gcc -o gentone gentone.c -lm
./gentone busy 480 620
Wavelength 1 (in samples): 16.7
Minimum samples (1): 50 (3.00.3 wavelengths)
Wavelength 1 (in samples): 12.90323
Minimum samples
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ACK
2007/4/12, Razza [EMAIL PROTECTED]:
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--
Alberto
On Thu, Apr 12, 2007 at 10:25:37AM +0100, Joao Pereira wrote:
Hello
Im trying to install an old version of Asterisk.
But it isnt working:
when I run make install:
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
Hello ,
iam having 6 asterik cards on three different servers
I am using asterisk 1.2.14 with zaptel 1.2.12 on fedora core 5 (2.6.17.1).
now every 3 days i need to rmmod/modprobe wctdm driver to detect the call.
callers wont get the IVR prompt. after rmmod and modprobe the wctdm ..it
works fine.
Hi guys!
Im using Asterisk 1.2 with mISDN support.
I have problems with Pickup calls with my Grandstream Buttons . I set up on
Dial Plan this:
Exten = _**XXX,1,Pickup(SIP/{EXTEN:2}) but it doesnt work if the call
comes from mISDN. So, I wanna do something to this:
Exten =
Hello ,
iam having 6 asterik cards on three different servers
I am using asterisk 1.2.14 with zaptel 1.2.12 on fedora core 5 (2.6.17.1).
now every 3 days i need to rmmod/modprobe wctdm driver to detect the call.
callers wont get the IVR prompt. after rmmod and modprobe the wctdm ..it
works fine.
Hi,
I have to set call length to 3min, but before hangup have to warn
caller. There are many IVRmenu and submenu options with different
warning audio.
I have to measure somehow the audio file length and subtract it from 3
minutes.
exten =
Hello
Thanks a lot for the help.
I just commented these lines and its working:
#ifneq ($(wildcard
$(CROSS_COMPILE_TARGET)/usr/include/linux/ixjuser.h)$(wildcard
$(CROSS_COMPILE_TARGET)/usr/local/include/ixjuser.h),)
# CHANNEL_LIBS+=chan_phone.so
#endif
I just hope that this doesnt bring me
I confirm the same behaviour. I use asterisk with Mera Softswitch (with
SIP HIT).
After upgrading from 1.2.13 to 1.2.14 Maximum retries exceeded...
messages began to appear in logs. About 10% of calls were lost. I've
dumped such calls and don't see anything suspicous in Mera's packets.
Quoting Stephen Bosch [EMAIL PROTECTED]:
I'm trying to decide which phones to experiment with. I have these
options:
- A combination of Polycom, Aastra and Snom
- Just Polycom
One the one hand, I'd like to keep things uniform, since it greatly
simplifies provisioning. On the other hand, I
Hello to everybody, I have a problem with the disposition filed that
asterisk write in mysql table.
What I notice is that for every outbound calls (for example to a mobile
phone) I see in disposition field the string ANSWERED when I reject the
call and also when I really answer the call, while
You hit the nail on the head.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906
-Original Message-
I think this has to do with how your dial plan is setup. If you are making
a call to a cell phone, i'm assuming that you are using an FXO (or some sort
of phone service). My guess is that the disposition is being marked
ANSWERED because the FXO is picking up (or the phone service is) and
I blogged about it here
http://deancollinsblog.blogspot.com/2007/04/software-patents.html
Though I think GigaOm nailed it when they wrote
Verizon can't make the Internet go away with a patent lawsuit.
http://gigaom.com/2007/04/08/voip-patent-mess/
Cheers,
Dean Collins
Stephen Bosch wrote:
Lee Jenkins wrote:
Hi all,
I just purchased a Polycom 301 for my home office and I believe I have
it setup correctly as I can dial out, receive calls in, etc. However,
I'm having the following issue:
When calling a local number over a Zap line, I hear a lot of feed back
No luck yet. No response from Digium support so I guess that they are still
waiting for the Zaptel test code.
Greg
-Original Message-
From: Stephen Bosch [mailto:[EMAIL PROTECTED]
Sent: Thursday, 12 April 2007 4:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
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Hello
Thanks a lot for your reply.
Im now using asterisk-1.2.10 and the problem disappeared.
Thanks
regards
Joao Pereira
Edoardo Serra wrote:
Same to me !!
Calls from OpenSER to Asterisk
It happens only with Asterisk versions = 1.2.14
I'm going to capture some traffic
Tnx for help
Regards
You seem to have misplaced your message/comment/question.
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In article [EMAIL PROTECTED],
Gordon Henderson [EMAIL PROTECTED] wrote:
On Wed, 11 Apr 2007, Tony Mountifield wrote:
Alejandro Mejía [EMAIL PROTECTED] wrote:
I would like to know how to playback an audio file to the caller, and while
it's played asterisk to continue executing the next
moises, guys,
just an update, steve released new packages early april.
i just did a successful compile, tomorrow i will test with a live e1 line.
i managed to compile it with asterisk-1.4.2
a series of patches is on the way after a successful test.
[EMAIL PROTECTED] wrote:
nivlekch, nice to
On Wed, Apr 11, 2007 at 07:32:52PM +0300, Dovid B wrote:
I wrote this ages ago. You may want to get more current software than the
URL's that are listed.
#YUM INSTALLS
yum -y install gcc
yum -y install kernel-source
actually: kernel-devel (or kernel-smp-devel)
yum -y install bison
yum
Exactly. It's a weird issue, and I can't imagine what the problem is,
except maybe for bad phones (but then again, why would the phones be only
bad with 2.x?)
UnlessI have bootrom 3.2.2.0019. Is that what people running thelatest
have?
Mike
-Original Message-
From: [EMAIL
Stephen Bosch wrote:
Stephen Bosch wrote:
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
I'm trying to decide which phones to experiment with. I have these options:
- A combination of Polycom, Aastra and
Hi Nivlekch,
Thanks for that, just a comment:
What do you mean by new packages? new for spandsp, libmfcr2, unicall?
chan_unicall?
On 4/12/07, nivlekch [EMAIL PROTECTED] wrote:
moises, guys,
just an update, steve released new packages early april.
i just did a successful compile, tomorrow i
Somehow, I ended up with BootROM 3.2.3.0002 (which as far as I can tell hasn't
been released yet...) and SIP version 2.1.0.2708.
I do see the sluggish buttons from time to time. Rarely, but I do see it.
--TS
Mike [EMAIL PROTECTED] 4/12/2007 9:59 AM
Exactly. It's a weird issue, and I
Hi.
I'm stuck into an odd situation.
Here's what happens:
4 Thomson ST2030S
2 Cisco 7912
3 Cisco 7940
2 AAstra 480i
Asterisk 1.2.17
Diva 4BRI + chan_capi
I've just upgraded yesterday from Asterisk 1.2.13 to 1.2.17.
Until yesterday, everything was just fine with 1.2.13.
Immediately after the
also I've seen that not having the correct version of sip.cfg and
phone1.cfg could cause weird problems. Make sure you are using the ones
that came with the firmware.
Mike wrote:
Exactly. It's a weird issue, and I can't imagine what the problem is,
except maybe for bad phones (but then
Hi,
Let me join all of you, interested in such monitoring tool.
Cheers
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Dovid B wrote:
I wrote this ages ago. You may want to get more current software than
the URL's that are listed.
I just changed the version numbers before doing the script ;)
--
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Lee
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Hi all,
I have asterisk 1.2.13 and problem is about DTMF.When i have incoming call on
Toll-Free number asterisk accept DTMF digits but dial only first in context.
Per instance:
When i press 1 it is OK,but when i try to dial extension 700 asterisk dial only
first digit(1) and i receive from
On Wed, 11 Apr 2007, Kevin P. Fleming wrote:
Alan Ferrency wrote:
This means that all queue activity is associated with a SIP channel
in the logs, which is not acceptable.
This is why we added the 'membername' argument to the
AddQueueMember application, so that queue logs can reflect a
Hi All,
I have 2 GXV-3000 phones. Working fine when I manually call the phones.
However, if I use a call file to initiate my call to phone 1, then the
dial plan calls
the second phone only the second phone shows video not the first phone.
How can I get video showing on the first phone also?
Hi all,
Is it possible to configure an extension number to dial a sip address?
For example:
exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED])
That way I can dial to a sip name using my Hardphone that is not able
to dial using names just numbers.
Thanks in advance.
Ronaldo.
(I hope putting my sip
Alberto Pastore wrote:
Firmware on 7940 is 8.6 (the latest one).
I had the same issue. I ended up moving back to firmware P0S3-07-4-00
on the phone. I did a telnet into the phone, did a show register and
shaw some very weird info. Normally, I would see:
LINE REGISTRATION TABLE
Proxy
On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote:
Hi,
I have to set call length to 3min, but before hangup have to warn
caller. There are many IVRmenu and submenu options with different
warning audio.
I have to measure somehow the audio file length and subtract it from 3
minutes.
I
Hi,
My asterisk was working fine but today my calls won't out of my asterisk box.
Restarting asterisk i need to wait around 10 min to can run sip show
registry command.
If i try to run this command before, i receive a error like: no such command.
Why this happen ?
Thanks.
--
Frederico
I found *something*. I've gone into my CPU graph (on the phone, in status -
diagnostic). Two phones, one running 1.6.7 and one running 2.1.0, both on
the same Hub, with the same general configuration (different SIP
registration, and each using it's version-specific sip.cfg file).
The pre-2.x
Hi Frederico,
I sometimes have the same problem tooI think the problem is related
to VoIP providers registrations. Are you using VoIP services on your PBX?
Thank you.
Giorgio Incantalupo
Frederico Madeira wrote:
Hi,
My asterisk was working fine but today my calls won't out of my
I'll be sending Digium support the info they requested later today. I
hope it helps.
Greg Siemon wrote:
No luck yet. No response from Digium support so I guess that they are still
waiting for the Zaptel test code.
Greg
-Original Message-
From: Stephen Bosch [mailto:[EMAIL
I'm currently looking to interconnect my Asterisk PBX system with the PSTN
via a digital PRI/T1.
I know a multitude of options exist for internal PCI cards
(Digium/Sangoma/Rhino), I was wondering if anyone has any experience or
recommendations of external PRI media gateways that support SIP.
So
Mike wrote:
I found *something*. I've gone into my CPU graph (on the phone, in status -
diagnostic). Two phones, one running 1.6.7 and one running 2.1.0, both on
the same Hub, with the same general configuration (different SIP
registration, and each using it's version-specific sip.cfg file).
May i ask why not internal?
On 4/12/07, jameson asterisk [EMAIL PROTECTED] wrote:
I'm currently looking to interconnect my Asterisk PBX system with the PSTN
via a digital PRI/T1.
I know a multitude of options exist for internal PCI cards
(Digium/Sangoma/Rhino), I was wondering if anyone has any
Can anyone recommend software that will allow me to utilize my VoIP
provider and send fax over IP?
I use Asterisk now for my phone system.
Thanks!
Wiley E. Siler
Director of Information Technology
Education 2020
4110 N. Scottsdale Rd. Ste 110
Scottsdale, Arizona 85251
(480) 296.0260
Playing with hints/presence/BLF on asterisk I've made the following
discoveries.
1. The wiki at http://www.voip-info.org/wiki/view/Asterisk+presence says:
If you add incominglimit=1 to your peer in sip.conf, the SIP
channel will notify you when that extension is busy.
As
Hi,
I'm having problems installing codec g729 on my Asterisk that's running on
FreeBSD 6.0
codec_g729a.so module loads ok, but the register utility doesn't seem to
register the license key correctly, because when I issue show g729 under
Asterisk's CLI it says that the command is invalid.
It
That's just the thing. There are manifold options, but they are all quite
expensive.
--
Alex Balashov [EMAIL PROTECTED]
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On Thu, 12 Apr 2007, Wiley Siler said something to this effect:
Can anyone recommend software that will allow me to utilize my VoIP
provider and send fax over IP?
Asterisk can send faxes, if you make it interoperate with a few
well-known open-source utilities and/or software packages,
Drew Gibson wrote:
Stephen Bosch wrote:
Stephen Bosch wrote:
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
I'm trying to decide which phones to experiment with. I have these options:
- A
Bob Smither wrote:
On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote:
Hi,
I have to set call length to 3min, but before hangup have to warn
caller. There are many IVRmenu and submenu options with different
warning audio.
I have to measure somehow the audio file length and subtract it
Lee Jenkins wrote:
Stephen Bosch wrote:
Sidetone can be set in the phone configuration; before you do that,
though, I need to know what you mean by feedback.
Sorry, should have been more detailed. It's a sort of background
humming noise, almost like that if you placed the phone next to a
Drew Gibson wrote:
We have Cisco, Aastra 480i and Grandstream GXP2000 phones in house.
I only recommend the Cisco phones to people I don't like, overpriced and
far too much work.
The Aastra 480i is a good quality phone, on par with Cisco and probably
with Polycom (though I've never used
On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote:
Hi,
I have to set call length to 3min, but before hangup have to warn
caller. There are many IVRmenu and submenu options with different
warning audio.
I have to measure somehow the audio file length and subtract it from 3
minutes.
I
On Thu, Apr 12, 2007 at 11:59:00AM -0400, jameson asterisk wrote:
I'm currently looking to interconnect my Asterisk PBX system with the PSTN
via a digital PRI/T1.
I know a multitude of options exist for internal PCI cards
(Digium/Sangoma/Rhino), I was wondering if anyone has any experience or
Wiley Siler wrote:
Can anyone recommend software that will allow me to utilize my VoIP
provider and send fax over IP?
No, but I can recommend that you read this to see why you shouldn't bother:
http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
Lee.
Doug Lytle ha scritto:
Alberto Pastore wrote:
Firmware on 7940 is 8.6 (the latest one).
I had the same issue. I ended up moving back to firmware P0S3-07-4-00
on the phone. I did a telnet into the phone, did a show register and
shaw some very weird info. Normally, I would see:
...
But
Basically, I want to send bulk faxes to a list of my clients.
It is time consuming for a person to individually fax so a blast type
solution seems best.
Over IP is of course to save money...
Thanks for the link, reading now...
Any suggestions for the blast then?
Wiley E. Siler
Director of
Hello Francis,
I also hev asterisk and sipura. Can we chat online on gmail/yahoo. Let's
make some experiments... I hev the same problem like you.
On 4/12/07, Francis Augusto Medeiros [EMAIL PROTECTED] wrote:
On 10 de abr de 2007, at 23:05, James Harper wrote:
2 - How can I gain full
The Cisco phones are quite good. The thing that most people don't tend to
appreciate about them is that they all are designed essentially for
mass-provisioning in large environments, and to operate with Call Manager.
Provisioning them using their GUI/configuration interface on a one-off
From overall apprecation feedback :
#1 Polycom (Any)
#2 Aastra 480i
#3 Cisco 7940+
#4 Linksys SPA-94x
On 4/11/07, Stephen Bosch [EMAIL PROTECTED] wrote:
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
I'm
Mike,
Got off the phone with Polycom on this I have the same problem
with my new 601 phone here (haven't seen the problem on the 650).
I'm trying to find answers and Polycom's only got one reported case
of this (which I find bazaar, but whatever). The problem was
resolved, the
Stephen Bosch wrote:
Lee Jenkins wrote:
Stephen Bosch wrote:
Sidetone can be set in the phone configuration; before you do that,
though, I need to know what you mean by feedback.
Sorry, should have been more detailed. It's a sort of background
humming noise, almost like that if you placed
Noah,
I am just using a dlink router for dhcp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Wednesday, April 11, 2007 2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom - Static
Dear Jason,
Here in my company we use an applet it java IAX, and it functions very well!
If to want to visit the URL is http://www.virgos.com.br, calls the service
as 0800Web.
Leonardo Silva
2007/4/5, Jason Wolfe [EMAIL PROTECTED]:
I need to decide on the best way to add a voip SIP or IAX
Eric ManxPower Wieling wrote:
I'll be sending Digium support the info they requested later today. I
hope it helps.
We have a developer working on extending Zaptel to support pre-echo
audio capture right now, so that we can work on debugging these issues
with real data instead of just
People, I have a Debian box with Asterisk and I've installed the Destar
package in order to get web managing of my voip system.
After I installed Destar, it runs on localhost:8080, but my server
does not have X-Window to access to it so I can engter the web interface..
So how can I change
Does anyone know who maintains the Asterisk-java web site at
asterisk-java.org? The site seems to have been unavailable for a couple
of days now.
Doug
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To
Either analog modems or a PRI, and Hylafax for automation, no VOIP
involved there.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, April 12, 2007 10:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Wiley Siler wrote:
Basically, I want to send bulk faxes to a list of my clients.
It is time consuming for a person to individually fax so a blast type
solution seems best.
Over IP is of course to save money...
Thanks for the link, reading now...
Any suggestions for the blast then?
My
Alberto Pastore wrote:
But why does 8.6 seem to work with previous asterisk 1.2.13??
That I wouldn't be able to answer.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
Hi,
When I tried to use speex (8 khz) codec I got
following warning messages on the Asterisk console.
The other end was pjsip and I was testing this in
local network.
Here is a exact message:
WARNING[6055]: codec_speex.c:237 speextolin_framein:
Out of buffer space
Has anybody had success in
Jessee J Holmes wrote:
Got off the phone with Polycom on this I have the same problem with
my new 601 phone here (haven't seen the problem on the 650).
I am using an IP650 with the latest firmware (and the corresponding
sip.cfg file) and I have seen this behavior. It is most noticeable
Drew Gibson wrote:
The Aastra 480i is a good quality phone, on par with Cisco and probably
with Polycom (though I've never used them). Voice quality is good,
phone
feels robust. Config is well documented and contained in two text files
(one global, one MAC specific). Good web interface
Hi there list!
I want to catch all numbers that don't exist, play a
nice message and restart operator, this is different
from dial i because that is for incorrect extensions,
an undefined number will give a busy signal, something
I don't like
You can search for the word irc to see my comments,
Hum, I know Stefan, he is an asterisk-java dev, but he is not online
right now, I will let him know ASAP. Thanks!
On 4/12/07, Doug Garstang [EMAIL PROTECTED] wrote:
Does anyone know who maintains the Asterisk-java web site at
asterisk-java.org? The site seems to have been unavailable for a
Thanks all... Looks like I will have to let them know that FOIP is a no
go and that we can automate on Asterisk though...
Thanks!
Wiley E. Siler
Director of Information Technology
Education 2020
4110 N. Scottsdale Rd. Ste 110
Scottsdale, Arizona 85251
(480) 296.0260
(866) 320.2083 ext. 1003
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello Asterisk Gurus!
I have a very simple question. I've just started playing around with
Asterisk and BSD box. I also have grandstream ip phone and installed
asterisk from ports. Now I'm on my very first steps to configure
Asterisk. The question
Hey guys,
What are some of the numbers you guys want graphed?
Anything that is a number, or any kind of information.
Now I have
Agents logged in and out
# of queues
total calls
total channels
What else?
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Hi,
It's really a simple question!
I've just started playing with asterisk too, and I think what you want
could be found in the 4th chapter of Asterisk: The Future of the
Internet. It's a open book you can download from
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11.
I hope
On 4/12/07, Brandon Kruse [EMAIL PROTECTED] wrote:
Hey guys,
What are some of the numbers you guys want graphed?
Curious how you are going to do this and will it be backwards portable. One
of our engineers wrote an app that queries the manager interface to build
RRD data. That's sent over
Hello eveybody,
I've been looking for a way to share trunks between two asterisk
servers. I guest I have to use Dundi, but I've not found the exact
method yet. I need a way to allow users registered in one server to
use the another server's trunks in the case the first server's trunks
were busy
On Thu, 12 Apr 2007, Edgar Guadamuz said something to this effect:
I've been looking for a way to share trunks between two asterisk servers.
Provided that the Asterisk servers can be set up to hold identical
SIP contacts (URIs), you can just set up a dialplan such that it fails
over if a
I've had experience with quite a few different phones, so I think I'm
qualified to drop my two cents:
Alex is quite right that the Cisco phones are only designed to be used
with Cisco Call Manager. They are capable of being decent SIP
telephones, but Cisco won't provide the documentation so that
Another way is to run the calls through a SIP proxy such as SER which can
hunt through two Asterisk UA endpoints, depending on a variety of
parameters including failure at a primary and fallback to a secondary.
--
Alex Balashov [EMAIL PROTECTED]
Ken,
You have certainly had experience with a broader range of phones, so I have
no doubt you can lend more insight on this count.
But for what it's worth, my experience is largely confined to the Cisco
7960s. I've never had any trouble getting any SIP firmware image to
register with
On Fri, 2006-09-29 at 11:12 -0500, Pato Valarezo wrote:
Hi, sorry for the question, i've been searching for a real time billing
system for asterisk with zap/sip support, for use in post paid systems
like locutorios, do you know of or use any ?
Give a try to StarshopOSS:
Thank you Alex and It would be possible to do that using IAX too,
wouldn't it?
I mean something like
exten=_9NXX,1,Dial(Zap/g0/${EXTEN:1})
exten=_9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
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I hope I don't get flamed the first time I post to a new list. I have
spent a couple of hours poking around without seeing anything like this.
The problem is, as soon as I load the Zaptel drivers (with a TDM-31B
card), ssh into or out of the server is broken. Trying to ssh in, I get:
Has anyone tried to pass sccp through a cheap router / nat box?
I have gotten sccp to go through a cisco pix just fine, but I can't seem
to get it to go through a ipfilter box or a basic netgear / linksys
router. I was under the impression that sccp was a lot more nat
friendly, but at the
Hi All
Has anyone managed to get Asterisk 1.2 faxes working reliably with
spandsp 0.0.3? I am running Asterisk 1.2.17 and spandsp 0.0.3pre28 with
a Digium b410p card. Everything compiled smoothly but only about 70% of
faxes come through ok. Debugging shows nothing more than: app_rxfax.c:
Hi there,
I am new to this ML. Recently I started working on Asterisk 1.4 + RAGI +
Ruby on Rails to create a call history browser.
To record call history, I am trying to capture dialup, answer and hangup
events. To check what status a call is, I use channel_status() that RAGI
provides.
I am
Certainly. Any signaling / trunking protocol will do, in principle.
On Thu, 12 Apr 2007, Edgar Guadamuz said something to this effect:
Thank you Alex and It would be possible to do that using IAX too,
wouldn't it?
I mean something like
exten=_9NXX,1,Dial(Zap/g0/${EXTEN:1})
Victor Hoodicoff wrote:
I think your impressions of Aastra are outdated. Install the latest
firmware, download the latest documentation and test and THEN give an
opinion!
Did you miss the part when I wrote I have Asstras sitting on my desk
collecting dust. I program on average about 5 per
I have been trying to setup a PAP2 adapter on a remote network but can't
seem to get it to work. The unit will register with the server and it can
make calls to extensions on the Asterisk server but it can't receive any
calls and it can't make any calls outside of the Asterisk server.
I also
Greg Woods wrote:
I hope I don't get flamed the first time I post to a new list. I have
spent a couple of hours poking around without seeing anything like this.
The problem is, as soon as I load the Zaptel drivers (with a TDM-31B
card), ssh into or out of the server is broken. Trying to ssh in,
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