We have found a new thing called 'the pub'
It even provides beverages.
Trust me, you can't find a program that can do that!
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergio Veltri
Sent: Wednesday, 20 April 2005 3:37 AM
To:
OK, been messing with RealTime like a week off and on, I can safely say it's
killing me!
I have dug and dug and dug to find what I am missing, no dice.
I am running the latest version of * from CVS as of about a week ago.
Call comes in from a PRI into the todd_test_1 extension, if I uncomment
Guys.
Ive read on the wiki that a common problem with nat is that you can only
have 1 sip phone behind, how do you get around this issue? Having a sip
enabled router behind the nat like the GS 488 489 or 486? Or how have you
done it without having any kind of linux box (SER or *) behind the nat.
Is there any way of sending text messages to SIP ATA's or phones? Like SMS
but for SIP IAX2 ATAs or phones.
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How have you done it for */* combinations?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Martes, 19 de Abril de 2005 07:35 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Using voicemail
I have been looking and have tried many many things but not
have been able to get it working
I am running Connected to Asterisk
CVS-HEAD-04/14/05-11:56:07 currently running on localhost (pid
= 1927)
Regards
Paul Dracevich
Wireless Technology Consultant
Wayby Group
Mobile +64 29
On Tue, Apr 19, 2005 at 16:44:58 +0100, Chris Hills wrote:
Ronald Wiplinger wrote:
I have IPv6 (via tunnel) available.
Is there a solution for IPv6 available?
Hi Ronald
This is something I would like as well. Unfortunately there is no
support for IPv6 at present. Perhaps you could
On 4/3/05, Ronald Wiplinger [EMAIL PROTECTED] wrote:
You can't see the sweat, but ...
I would like tp post my improvements to ASTCC somewhere, ... but where???
Post them as patches to bugs.digium.com and then they can be
incorperated into the main code.
Julian J. M. wrote:
I want asterisk to receive incoming faxes (via rxfax application) and
send them by mail. The problem is that, although the fax machine and
the asterisk log report a succesful transfer, the tiff file is just
I have not experienced this before, but I am using spandsp-0.0.2pre10,
I'm using asterisk-oh323-0.6.5 with the Janus patch 4 versions of
pwlib (v1.6.6.3) and openh323 (v1.13.5.3), and using it to connect
to my provider's switch.
The effect that I am seeing is that a call starts off fine, but suddenly
after a few minutes the audio coming into Asterisk via OH323 gets
Hi :)
When I send an incoming call to a queue, I'm doing this:
exten = 6608140,1,SetCallerID(CCUK)
exten = 6608140,2,SetCIDName(CCUK)
exten = 6608140,3,Queue(ccuk,r)
I want the phone to say 'CCUK' - the queue name is more important to know than
the incoming Caller ID :)
Unfortunately the SIP
I like the way FWD does to connect to other friendly networks via
**, however, I am not sure how is the best way.
Can I just use
exten = **393.,1
exten = **394.,n ...
???
bye
Ronald
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-- Starting simple switch on 'Zap/3-1'
-- Executing NoOp(Zap/3-1, 9229443944-) in new stack
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Zapateller(Zap/3-1, ) in new stack
-- Executing BackGround(Zap/3-1, custom/Welcome) in new stack
-- Playing 'custom/Welcome'
Gavin Hamill wrote:
Hi :)
When I send an incoming call to a queue, I'm doing this:
exten = 6608140,1,SetCallerID(CCUK)
exten = 6608140,2,SetCIDName(CCUK)
exten = 6608140,3,Queue(ccuk,r)
I want the phone to say 'CCUK' - the queue name is more important to know than
the incoming Caller ID :)
Hi,
I have a Cisco ATA 186 that I bought on my recent overseas trip and its
the I2 series which has higher impedance than the New Zealand standard
600ohm.
Is there something I can do to make it listen to my DTMF tones?
Regards,
Sahil Gupta
VoiceValley
On Wednesday 20 April 2005 10:32, Ronald Wiplinger wrote:
So my question is, how can I change the sip username from
sip:[EMAIL PROTECTED] to sip:[EMAIL PROTECTED] ?
Shouldn't be there a quote mark and two values, like:
SetCallerID(Ronald 123456789)
Just tried a few combinations of that,
Me wrote:
OK, been messing with RealTime like a week off and on, I can safely
say it's killing me!
I have dug and dug and dug to find what I am missing, no dice.
I am running the latest version of * from CVS as of about a week ago.
Call comes in from a PRI into the todd_test_1 extension, if I
In your incoming context add
exten = fax,1,Goto(fax,2202,1)
On Wed, 2005-04-20 at 12:26, Ronald Wiplinger wrote:
-- Starting simple switch on 'Zap/3-1'
-- Executing NoOp(Zap/3-1, 9229443944-) in new stack
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing
U don't need to have sound device for * sound service running
just make sure that you have in modules.conf
noload = chan_alsa.so
noload = chan_oss.so
On Wed, 2005-04-20 at 08:44, [EMAIL PROTECTED] wrote:
Hi,
I installed asterisk-1-0-7 and running it succesfully. But iam unable to use
the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
John Riek wrote:
| I would like to use Asterisk as a standalone voicemail server and
| integrate it with a Cisco Call Manager PBX. I need to know how to
| run the voicemail system independently. Does anybody know how to
| do this?
|
What Version of
Hi there
I have got a really strange issue and my problem is not that
it is not working, but why it is working.
I have Asterisk set up on a public IP, but the clients are
behind a Port Restricted NAT with no support for UPnP. My clients dial into a
meetme conference. When I don't
Try using
SetCIDNum(CCUK)
-Arun
On 4/20/05, Gavin Hamill [EMAIL PROTECTED] wrote:
On Wednesday 20 April 2005 10:32, Ronald Wiplinger wrote:
So my question is, how can I change the sip username from
sip:[EMAIL PROTECTED] to sip:[EMAIL PROTECTED] ?
Shouldn't be there a quote mark and
Vladyslav wrote:
In your incoming context add
exten = fax,1,Goto(fax,2202,1)
It did not work ;-(
[incoming_88097680]
exten = s,1,NoOp(${CALLERIDNUM})
exten = s,2,Wait(1)
exten = s,3,SetCallerId(9${CALLERIDNUM})
exten = s,4,GotoIfTime(08:00-21:20|sun-sat|*|*?house-day,s,1)
exten =
Hi,
I was just wondering about a comment I found in the voip-info.org wiki:
The DIGIUM TE410 PRI card, requires a motherboard with a 64bit 3.3v PCI slot.
Given the bandwidth requirements, it would be better to have a 133Mhz slot if
available.
Since the card seems to always clock at 33MHz. I
Hi guys,
Is it possible turn on/off VAD (silence
suspression) w/ Asterisk?
Thanks in advance :),
Pavel
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Hi all, has anyone else had any experience with these new Snom devices.
I am having trouble with placing calls on hold. The hold works fine, the
music on hold kicks in, but when you take it off of hold the voice goes
really choppy. I can't tell if it is a server side issue or not but I am
running
Steven Langley wrote:
I tried putting in nat=yes in the sip.conf file, and asterisk then rewrites
the sip message with the IP of the Nat and the external port. It still
works, but only if there is a constant flow of rtp traffic. If there is a
break in the traffic, then the connection is lost.
The Sipura SPA-841 has everything except memory buttons but has a directory
and speeddials so I don't think that's so important. Cheap and well made,
although if the speaker phone is very important, get Polycoms, it's the
business they are best in.
Chris Mason
www.anguillaguide.com
Pavel Siderov wrote:
Is it possible turn on/off VAD (silence suspression) w/ Asterisk?
Asterisk does not support VAD so it doesn't make sense to be able to
disable it.
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On Wednesday 20 April 2005 11:15, Arunachala wrote:
Try using
SetCIDNum(CCUK)
Nope, the most I can ever extract from any combination of the three 'CID'
commands is this in the SIP messages :(
From: CCUK sip:[EMAIL PROTECTED]
Cheers,
Gavin.
___
Yeah... unbelieveable but true: spam, defined often us undesired bulk
mail comes in many forms, including messages from this list.
I tried several times - obviously unsuccessfully, as you can see - to
unsubscribe from this list, and sice that did not work, i set my mail
server to BOUNCE list
Hi,
Has anyone used Cisco 2800 Integrated services router to intiate SIP
call to Asterisk. I would like to use it as gateway on to which T1
terminates and make Asterisk as my session target for few lines.
Please let me know if there are any issues.
Thanks,
Sharath
I have been looking and have tried many many things but not
have been able to get it working
I am running Connected to Asterisk
CVS-HEAD-04/14/05-11:56:07 currently running on localhost (pid = 1927)
Regards
Paul Dracevich
Wireless Technology Consultant
Wayby Group
Mobile +64 29
Anton Krall wrote:
Guys.
Ive read on the wiki that a common problem with nat is that you can only
have 1 sip phone behind, how do you get around this issue? Having a sip
enabled router behind the nat like the GS 488 489 or 486? Or how have you
done it without having any kind of linux box (SER or
Al wrote:
Yeah... unbelieveable but true: spam, defined often us undesired bulk
mail comes in many forms, including messages from this list.
You receive messages from this list because YOU signed up for it.
If you knew anything about email (or mailing lists), you could have
looked in the email
On Wednesday 20 April 2005 11:55, Arunachala wrote:
Hi Gavin,
Just went through the code. There is a check in the code to check
whether the CIDNum is a phone number (0-9,#,*) or no. If it is not a
phone number, it is replaced with the default CIDNum asterisk.
Hm, really smart :) If the SIP
Al wrote:
Yeah... unbelieveable but true: spam, defined often us undesired bulk
mail comes in many forms, including messages from this list.
I tried several times - obviously unsuccessfully, as you can see - to
unsubscribe from this list, and sice that did not work, i set my mail
server to BOUNCE
To breifly recap
Your main asterisk box runs linux, asterisk, ASTCC and MySQL
Another box runs linux, mysql, apache
The two sql servers are joined, updating each other?
or have I missed something?
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I've been running into something similar. The fax detect works reliably
to auto transfer the call. I see it Goto the new context, but instead of
actually ringing the fax extension, it just fails over as though it's
busy or something. I can always manually dial to the fax extension with
Paul Dracevich wrote:
I have been looking and have tried many many things but not have been
able to get it working
I am running Connected to Asterisk CVS-HEAD-04/14/05-11:56:07
currently running on localhost (pid = 1927)
With the ingredients you provide you can earn at most an answer
I'm sure this has been debated before, I'd like to get peoples input. I see
the hard drive as the single most likely point of failure on an * PBX. How
reasonable would it be to run the OS and config files from a CF card, mount
the /var/partition on a hard drive for the CDR recortds, logs and
It works for me with the default SIP settings. I am using the latest
firmware.
I have found that you have to restart * for it to pick up on the new PIN
code however.
On Wed, 2005-04-20 at 01:58, Master Abi wrote:
if I have conf = 80,111 in meetme.conf, I dial 80# and connect to the
conference,
When at the CLI, show channels shows nothing.
Look for ztcfg
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Hi,
The card need only 3.3V pci slot.
133MHz pci slots are 3.3V.
On Wed, 2005-04-20 at 12:21 +0200, Tais M. Hansen wrote:
Hi,
I was just wondering about a comment I found in the voip-info.org wiki:
The DIGIUM TE410 PRI card, requires a motherboard with a 64bit 3.3v PCI slot.
Given the
Is it possible to make Asterisk to execute a task when a called party answeres?
Does the MGCP protocol include support for notificate when a call is answered?
I have one Adit 600 w/ 40 FXS lines. When a call is initiated from such line to
the
PSTN through our E1 EuroISDN, I would like the Adit
Hi List,
Hi!
1-Skype-like softphone for *. is there any?
None that I know of. But IAX isn't bad in most of the firewalled environments,
give it a try. It only has to get a udp socket open for an outbound connection
(may well be NAT-ed) and to receive the answer packets back.
2-Just do
Hi
I have just purchased a Rhino Channel Bank and am using it connected to
asterisk via a digium TE410P. I am having problems with connecting
phones to the channel bank.
I have channel one connected to a patch panel, a line adaptor
chonnecting the phone cord to the patch panel, and then the
You need to add that to context where you have BackGround application
running.
house-day and house-night I believe.
On Wed, 2005-04-20 at 13:16, Ronald Wiplinger wrote:
Vladyslav wrote:
In your incoming context add
exten = fax,1,Goto(fax,2202,1)
It did not work ;-(
Hi there, quick question about queues
(B
(BWhen calling a queue (which contains eg 4 extensions) it tells me what
(Bposition I am in the queue and then plays some music$B!D(Jthat is fine$B!D(J
(Bhowever, If there is no-one in the queue , it tells me that im first in line
(Band then plays
The latest Version 0.92 of IPSwitchBoard can connect to your MySQL database and
show you call records with filtering on extension and from- and to date. IPS
now also will check if theres a newer version available on start-up and
offer to start the download. The Configuration page has changed
On 4/7/05, Thomas Andrews [EMAIL PROTECTED] wrote:
On Thu, Apr 07, 2005 at 10:15:09AM +0200, Thomas Andrews wrote:
I have a Fritz! card set up to use capi, however when incoming calls to
the card are answered, asterisk segfaults.
Have you tried a make clean then make install in the
I don't know about channel banks, but when you go T1 to T1 device
with a cable, you need the RX/TX pairs cross connected. Do you have a
T1 crossover cable in play or a straight through?
On 4/20/05, Dan Goscomb [EMAIL PROTECTED] wrote:
Hi
I have just purchased a Rhino Channel Bank and am
certainly do... and asterisk and the rhino see each other...
On Wed, 2005-04-20 at 08:38 -0400, BJ Weschke wrote:
I don't know about channel banks, but when you go T1 to T1 device
with a cable, you need the RX/TX pairs cross connected. Do you have a
T1 crossover cable in play or a straight
Yes, same problem here. Sign-ed up with VoipJet and seems to work
just fine (prices for most areas we call are cheaper too from what I
saw). Only been using them for 24 hours so can't say much about
long-term stability, but so far so good.
Pedro
On 4/19/05, Matthew Asham [EMAIL PROTECTED]
Hi All
Can anyone help with this message?
We are using a Swissvoice with G729 on the latest CVS of Asterisk
Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid
data (4 bytes at the end)
Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid
data (4
- Original Message -
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, April 20, 2005 4:42 AM
Subject: Re: [Asterisk-Users] RealTime ignoring switch=
Realtime/[EMAIL PROTECTED]
Me
On Thu, Apr 07, 2005 at 10:15:09AM +0200, Thomas Andrews wrote:
I have a Fritz! card set up to use capi, however when incoming calls
to
the card are answered, asterisk segfaults.
Have you tried a make clean then make install in the chan_capi source
directory make sure the header
Henry Devito wrote:
I am already doing this with AGI, PERL, and PHP to set up the page
groups. I will release the code as open source if people are
interested. I'm not the best PERL scripter in the world but it works.
Attached is the agi I'm using. This is a modified script from a post on
On Tue, Apr 19, 2005 at 06:24:09PM -0700, trixter http://www.0xdecafbad.com
said:
as a whole. I enjoy cheap computers, if it were not for microsoft
creating windows, making computers easier to use for everyone, the mass
production and highly competitive hardware market would not exist. If
Can you post your config's? What version of * are you using? This doesn't
(Bhappen on any of my queues. I have queues set up on several customers
(Bsystems. If there are agents/members available the caller rings them
(Bdirectly, no announcements played.
(B- Original Message -
-Original Message-From: Goutam Shaw
[mailto:[EMAIL PROTECTED]Sent: Tuesday, April 19, 2005 11:22
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] FXO lines on TDM04B not
responding
I ran
into the situation where 3 of the 4
My opinion is that both are Crap. Both of them have a flaw in their base
design, which is difficult to explain in a post like this. Suffice to
say that these two applications neither support nor designed for mutilpe
routes ( multiple Area codes with Destination groups) nor multiple rate
On Wed, 20 Apr 2005, Pedro wrote:
Yes, same problem here. Sign-ed up with VoipJet and seems to work
just fine (prices for most areas we call are cheaper too from what I
saw). Only been using them for 24 hours so can't say much about
long-term stability, but so far so good.
I had this
Does anyone here know of any general, good voip mailing list? I am
having a hard time with broadvoice and the company is not answering
its phone.
TIA,
GM
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You could work this out with setgroup checkgroup, and create 2 queues,
(Bif checkgroup jumps to 101+ (its the fifth caller) it goes to the
(Bsecond queue. Make sure that the only difference between the second
(Band first queue is the announcement.
(BUsing the above you will have to know in
On 4/20/05, Sean A. Newton [EMAIL PROTECTED] wrote:
Do the SetGroup and CheckGroup functions behavior differently in CVS-HEAD
vs CVS v1-0?
When I upgrade to CVS-HEAD my call waiting disable doesn't seem to work,
using:
exten = s,1,SetGroup(SIP${ARG1})
exten = s,2,CheckGroup(1)
exten =
it sounds like the default behaivor of an [EMAIL PROTECTED] setup.
(B
(Bnot that I am knocking [EMAIL PROTECTED] in anyway - its a great way to test
(Bnew features.
(B
(BOn 4/20/05, Henry Devito [EMAIL PROTECTED] wrote:
(B Can you post your config's? What version of * are you using? This
On 4/20/05, Al [EMAIL PROTECTED] wrote:
Yeah... unbelieveable but true: spam, defined often us undesired bulk
mail comes in many forms, including messages from this list.
I tried several times - obviously unsuccessfully, as you can see - to
unsubscribe from this list,
Obviously the problem
I would like to install G723.1 and G729 on an Athlon 64.
I looked at http://readytechnology.co.uk but I could not get a clue how
to compile / get all the things for an Athlon. It seems it is only for
Intel architecture, ...
Has anybody a clue how to get G723.1 and G729 on an Athlon 64 to work?
http://www.grandstream.com/y-gxp2000.htm
Looks like the phone is $139 from DigitNetworks.. Price looks good..
If anyone has one working with Asterisk, how does it sound/work?
Also, does it have caller ID with name? The Budgettones only support plain
old callerID number.. Very annoying!!
Thanks Vamsi for your feedback.
I would love to do it with Asterisk since I can do a lot more eventually.
I did try a couple of iax2 clients and I couldnt go past the FW in a
particular customer.
Thanks for your email.
Regards,
Sergio,
Date: Wed, 20 Apr 2005 09:28:24 +0530
From: Vamsi
The effect that I am seeing is that a call starts
off fine, but suddenly
after a few minutes the audio coming into Asterisk
via OH323 gets very
broken up to the point of being about 90% silence
with occasional brief
snippets of audio getting through.
hi,
any errors or warnings in Asterisk
On Wednesday 20 April 2005 13:56, Domjan Attila wrote:
I was just wondering about a comment I found in the voip-info.org wiki:
The DIGIUM TE410 PRI card, requires a motherboard with a 64bit 3.3v PCI
slot. Given the bandwidth requirements, it would be better to have a
133Mhz slot if
Ron,
Here is what I think.
Ready technology code will compile only with Intel IPPs. But there are
two options, either 1) you compile the codec using Intel IPPs to provide
the C and other base library functions, in which case you have to have
the Intel libraries and license available on each
Stefan,
Thanks for your feedback. I am testing everything to find the right
solution. It is an interesting project since the listeners will vary
everytime. Most of them are corporate users and thus unable to touch
the corporate FW.
I found a large international corporation that allows me to run
I have setup my system to give a company announcement if somebody calls, ...
I would like to avoid these announcements, if the caller is known by the
system.
Each caller I would like to put into a database with name. Now we know them!
If we know them, we do not announcement.
Is there anything
Dear All,
My boss has placed a requirement for me to forward all our IDD calls
through a partner's IDD service, which requires us to call a 8 digit
number, wait for 1 sec, before we send in the foreign number we're trying
to call.
As I can't find anything on getting the PBX to wait, i'm only
Here you have a sample that i used to test that agi was doing well.
#include stdio.h
main()
{
char line[80];
setlinebuf(stdout);
setlinebuf(stderr);
while (1)
{
fgets(line,80,stdin);
if ( strlen(line) = 1 )
{
Ronald Wiplinger napisa(a):
I would like to install G723.1 and G729 on an Athlon 64.
I looked at http://readytechnology.co.uk but I could not get a clue
how to compile / get all the things for an Athlon. It seems it is only
for Intel architecture, ...
Has anybody a clue how to get G723.1 and
I'm getting the same behavior, and can't seem to figure out where to
set it to act differently.
1.06 is the version I'm using.
I'm using AgentCallBack so my agents don't have to keep the line open
-- perhaps that has something to do with it?
I can't post my configs now (not at the office), but
I upgraded to CVS-HEAD-04/20/05-09:25:13 yesterday and I am now having
problems because Asterisk is not setting the language properly. My server
runs in Spanish so I use the SetLanguage option so my prompts are read from
the es directory inside the sounds directory. But now for some reason
I have a line dedicated to receive faxes. It basically answers, gives
you a prompt to dial 1 for fax, an extension or wait on the line for a fax tone.
After a few seconds it will timeout (using the t extension) and give the
user a fax tone. The problem is that if the user hangs up
Signate offers an interesting product they call 'webcall', which
basically contacts a client at a number they provide then connects
that person to a sales staff. Some potential for abuse but a nice idea
for support etc.
I know that it is possible to do (obviously) and well documented but
has
Chris Mason (Lists) wrote:
I'm sure this has been debated before, I'd like to get peoples input. I see
the hard drive as the single most likely point of failure on an * PBX. How
reasonable would it be to run the OS and config files from a CF card, mount
the /var/partition on a hard drive for the
I currently use an SPA-841 on my desk and don't have any problems with it
http://www.geekgazette.com/index.php?option=com_contenttask=viewid=24
I have been looking at these phones and they have more office features
http://www.zultystechnologies.com/index.jsp?tab=product_listtype=phones
-Kerry
Do the SetGroup and CheckGroup functions behavior differently in
CVS-HEAD
vs CVS v1-0?
When I upgrade to CVS-HEAD my call waiting disable doesn't seem to
work,
using:
exten = s,1,SetGroup(SIP${ARG1})
exten = s,2,CheckGroup(1)
exten = s,3,Dial(Sip/${ARG1},15,t)
Do you not need a
exten =
On Wednesday 20 April 2005 10:29 am, David Choo wrote:
Dear All,
My boss has placed a requirement for me to forward all our IDD calls
through a partner's IDD service, which requires us to call a 8 digit
number, wait for 1 sec, before we send in the foreign number we're trying
to call.
As I
So, here's my quandary:
1) Asterisk running CVS HEAD as of a couple days ago
2) Cisco 7960 SIP phones in a different subnet than the Asterisk server
3) NAT/Firewall device between 7960's and *
I can initiate a call from the 7960's just fine. They can call out
using our Broadvoice account and
Yes,
SIT messages and CLASS messages like...
Your selective call rejection service is now off
Your calls are being forwarded to XXX-XXX-
etc.
-- Mike
- Original Message -
From: jltaylor [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
When I place a call on my softphone to a external number the call is
placed, when I click transfer, dial internal extrention (e.g. 202)
then hit transfer again, the call is transfered to the 202 extention
fine.
However, when the other way Internal call comes in, extension 201
answers, and
I think the word crap is a pretty strong word and is not fare to the
authors. Everyone have their own requirements of how billing should or
should not work. Everyone is exposed to a different way a pre-paid calling
card platform should behave. I have been in pre-paid environment for almost
15
On Apr 19, 2005, at 9:12 AM, Mike Robinson wrote:
Yes, you CAN use your existing Meridian phones. There is a product
called a Handset Gateway that converts traditional digital PBX
telephones (Norstar, Meridian, Definity, NEC, etc) into SIP signaling
so
the existing phones and wiring can work
Thanks a lot ,
Make update is ok, But where i can check the version of my Asterisk ?
Obviously it is another simple one . :(
Date: Fri, 15 Apr 2005 22:01:28 -0400
From: Steve Totaro [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] a simple question .
To: Asterisk Users
I have a Uniden UIP200 behind a NAT and an * server behind another NAT.
I am able to register with * and place calls. However, once the call is
established, I cannot hear anything from either end (UIP200 as well as
the called destination). Then, I did the exact same thing with X-Lite
and
Hello. I checked in the wiki and read a bunch of old threads from this
mailing list but haven't found what I'm looking for.
I'm using a simple PHP script, and here is the relevant portion:
fputs($socket, Action: Monitor\r\n);
fputs($socket, Channel: Zap/1-1\r\n\r\n);
That works fine. As does
Thats exactly what I thought but there are many parts on the wiki where
they mention the more than 1 SIP client behind NAT mmyth. Oh well,
maybe an urban legend? :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent:
Try one w:
exten = _9001.,1,Dial(Zap/g1/64919669,,D(w${EXTEN:3}),) ***
exten = _9001.,n,Hangup()
Robert Andrew Keller
Ferndale School District #502
[EMAIL PROTECTED]
360-383-9228 PH.
360-383-9218 FAX
Paving the way for tomorrows genius.
From: David Choo [EMAIL PROTECTED]
Reply-To:
Hi Jairo,
Try with other values for the jitter in your Gateway (H323).
One customer have a scenario like this:
Phone/Fax Gateways H323 -with 16/8/2/1 Port FXS- --- GNUGK
--- Asterisk --- Zap (E1)
.. and only we need modified the jitter settings in two Gateways.
Rafael Gonzalez Lomeña
I have actually setup AstCC and got it working. I have found a couple problems
with it and I dont think the problems have anything to do with my setup. The
problems that I am seeing are:
1) Out of the box, the CDRs dont work. I have a quick document that explains
why and how to fix it. If you
On Wed, 20 Apr 2005 10:24:37 -0500
Josiah Bryan [EMAIL PROTECTED] wrote:
On Wednesday 20 April 2005 10:29 am, David Choo wrote:
Dear All,
My boss has placed a requirement for me to forward all
our IDD calls
through a partner's IDD service, which requires us to
call a 8 digit
number, wait for 1
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