When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why?
When a phone call isn't from queue then att transfer works fine.
In features conf I have *1 for recording, *2 for att transfer and #1 for blind.
In queue blind transfer works. For disconnect I have #0.
I guess that
No chance, as ar as I know.
Brooktrout cards are closed-source in all
senses.
To use four of them with Hylafax in my company we was
forced to buy Enterprise hylafax licenses from ifax.com.
Mimmus
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Hi Faisal,
I have the following in extensions.conf
exten = s,1,Answer()
exten = s,2,GotoIfTime(8:00-17:00|mon-fri|*|*?Ecn-incoming,s,1)
exten = s,3,GotoIfTime(17:01-7:59|mon-fri|*|*?Ecn-afterhours,5000,1)
exten = s,4,GotoIfTime(*|sat-sun|*|*?Ecn-afterhours,5000,1)
;exten =
hi,
after upgrade from 1.0.x to 1.2.x i cannot send faxes
my topology:
PSTN-wct4xxp-asterisk- -sip- ata (ht496,ht488,asus vp100) - samsung
sf2500 fax
log:
Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for
[EMAIL PROTECTED] - INVITE (With RTP)
Feb 13 23:50:35 DEBUG[27914]
thanks for the information Peter, its really helpful. Also Ihave one more question - do you have any idea how many such simultaneous calls can an asterisk server handle (say running od 2.6Ghz, 1GB Ram, fedora machine)?
Thanks,
Nitin
On 2/13/06, Peter Fern [EMAIL PROTECTED] wrote:
You can enable
I'm having trouble making calls over my VoIP provider. I do successfully
register, and when I try to establish a phone call Asterisk sends wrong
username and password. Instead of sending username and pass that I have
provided, he send username and pass of the SIP phone that is registered to *
You'll have to use uattended transfers for CCs.
l.
In data Tue, 14 Feb 2006 09:37:24 +0100, Tomislav Parčina
[EMAIL PROTECTED] ha scritto:
When agent tries to transfer a phone call (*2 - att transfer) he hangs
up. Why? When a phone call isn't from queue then att transfer works fine.
In
has anyone experienced a problem where RTP audio cuts out when doing
30-40 concurrent channels via sip?
The box is freebsd 6, asterisk 1.2.4, voip only (no libpri, no zaptel -
not even a timing source)
The box has plenty of bandwidth, when a call to the same box is iax2 it
works, but when its
Dear asterisk users,
I am presently playing with an [EMAIL PROTECTED] I am trying to find the
best codec solution for my voicemail records. I want to use ARI
(Asterisk Recording Interface) to read the messages. I first used the
default wav encoding that was not appropriate because my
On Saturday 11 February 2006 19:36, Steve Underwood wrote:
Matthew Fredrickson wrote:
On Feb 10, 2006, at 10:25 PM, Steve Underwood wrote:
Matthew Fredrickson wrote:
On Feb 10, 2006, at 1:21 PM, Gerard Saraber wrote:
DC in the signal through the echo canceller represents a signal the
Vincent Régnard wrote:
Could you indicate me the appropriate setting to use theses two files
format ?
From http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf:
format
The format setting selects audio file format(s) to use when storing
voice mail messages. The value is a string
this is not usefull for public enviroment. clients behind
nat does not work...
turby
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitin
GuptaSent: Tuesday, February 14, 2006 10:51 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
On Monday 13 Feb 2006 21:21, Chris Bagnall wrote:
Hello all,
I've started implementing iLBC on some of the ATAs we have floating around
clients' homes, but I'm coming against this error message with most of
them: codec_ilbc.c: Huh? An ilbc frame that isn't a multiple of 50 bytes
long from
I have a handful of uip 200 phones that are loosing registration.
the unidencom.txt file has 60 as the expire value.
in the sip.conf file and the context for each phone I have tried all 4
settings:
Tried to make the timeout larger that the uniden
expire=120
expirey=120
expires=120
Hi there. We're going to develop a call centre app for internal use in
our office.
The call centre is probably going to be a java-based client installed on
a windows machine that our secretary can use. Features should be a way
to see incoming calls, answer them, and then transfer the calls to
Arne Morten Johansen wrote:
Hi there. We're going to develop a call centre app for internal use in
our office.
The call centre is probably going to be a java-based client installed on
a windows machine that our secretary can use. Features should be a way
to see incoming calls, answer them,
Hi,
Im so confused regarding Telmex specification of my ISDN PRI line.
The following is the line specification from Telmex:
Phisical interface:g703
coding: hdb3
signalling:isdn pri
clock: recover
crc4: yes
framing:pcm30 or cas
For java based applications, I'd recommend
http://www.asteriskjava.org/latest/
- Original Message -
From: yusuf [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 14, 2006 10:34 AM
Subject: Re:
Ever since I migrated to 1.2.X, I've noticed a problem with the MOH
for any/all my queues.
When there is only one caller in the queue, the caller does NOT hear
the MOH. However, the moment more callers call into the queue, all
the OTHER callers will start hearing the MOH, but the first one
I 'm wondering ...
I have tried to use Asterisk external IP for some times ... but it never affects the VIA SIP Field
Is It normal ? When reading many books in SIP, this should be external IP, no ?
Did somebody manage to put/force the external IP in this VIA header ? If yes, how ? If not,
Hi,
I know that this topic has already been posted to
this list previously, but each time the list grows bigger it is more difficult
to find things.. Sorry to post this again then!
Does the message below mean that I would need 15+36
licenses?
lv09*CLI show g72915/36 encoders/decoders
of
Hi,
I am trying to register, two accounts of phoneserve in my asterisk box,
but I have had problems since he seems not to recognize the format
register = masteruser:[EMAIL PROTECTED]
I have one master account and two slaves accounts, in my ata 286 they
work well, but in the asterisk they have
On Feb 9, 2006, at 11:19 AM, Kevin P. Fleming wrote:
Steven Langley wrote:
I am using IAX2 softphones dialing into a meetme conference. In my
softphone
I was forcing uses to click on a button when they wanted to speak,
enabling
their microphone and disabling their speakers. This way when
Dan,
Thanks for quick reply. Do both Asterisk and Televantage
act as H.323 endpoints or does one of them act as a gateway? Also, do you have
any sample h323.conf files that I could use as a reference? Once again,
thanks for any help in advance.
Anish Basu
VOIP Engineer
Foniquity,
I'm trying to connect an Asterisk system to an Avaya Partner ACS R6
system. The problem I'm having is that I cannot get the partner system
to get CallerID over the T1 modlue. The partner is using the T1 with E
M signalling (which I don't think can be changed), and whatever I
tried didn't work. My
A 488 can mean a codec miss match. Check that your Asterisk box is
configured for g729.
Kurt
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
I noticed the issue today and came looking for confirmation when I came
upon this thread. My Grandstream does not have this problem.
SPA-941, Snom 320 and Aastra 480i all demonstrate this issue.
I'm going to
Lee Archer wrote:
Am I the only one with this problem? I've got Aastra phones running
Can Asterisk send RTP to a specific port number? For instance, I know I
can limit INCOMING RTP to certain ports in rtp.conf, but can I limit
OUTGOING RTP to a specific port (specifically port 5004 - I'm testing a
theory and need to be able to do this)
Thanks!
Hi All,
I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4
PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?,
other detail is: this card have 4 card green.
I need to know what is the best card for the following scenario: I need a
IVR for my
I am attempting to get a planet VIP-150T to register with
asterisk 1.2.4. After searching google Ive found what appear to be
instructions in German, Russian and Spanish. Has anyone perhaps seen this
before?
Asterisk is kicking back the following error:
Feb 14 09:59:32 NOTICE[21765]:
Hi,
I've just compiled asterisk and get
errors when trying to run it. Am I right in thinking that if my digium
card is not plugged into our ISDN line then this is normal. If I
replace zapata.conf with a blank zapata file asterisk runs fine. The
reason I ask is we don't yet have ISDN installed in
Hi,
I'm helping out with a political campaign and would like to use
asterisk to blast out about 200,000 calls with a short message from
the candidate.
Provider:
I'm thinking voipjet may be a good solution?
Hardware setup:
I will have access to several T-1 lines so I would just want to set up
Hi,
So I've done my research on Chanisavail, read the wiki, checked the
archive but can't seem to find anything to suit my scenario. I've
played around with it a lot, but I'm still scratching my head on what
I need to do.
What I need is to be able to accept a call by SIP and ring all
telephones
Luz,
You need red fxo daughter boards to terminate pstn lines, the green fxo
are for connecting analog extensions.
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Luz Lopez
Sent: Tuesday, 14 February 2006 10:04 AM
To:
On Tue, 2006-02-14 at 15:03 +, Luz Lopez wrote:
Hi All,
I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4
PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?,
other detail is: this card have 4 card green.
No - the green modules are
Dov Bigio wrote:
Does the message below mean that I would need 15+36 licenses?
lv09*CLI show g729
15/36 encoders/decoders of 50 licensed channels are currently in use
No. If you did, then you would have run out already, since you only have 50.
Each license gets you one encoder and one
Hi
I need guidance for trunking two asterisk servers
, I have two fixed IP address machines 192.168.20.5
and 192.168.20.10 , now I wanted the users of each
servers to dial to other server
I tried a lot was not successful ,
after a lot try with IAX I decided to try
If
Asterisk is in the public network, it will work. The problem is when Asterisk is
behind NAT and one of the client is also behind the same
NAT.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
turbySent: Tuesday, February 14, 2006 6:35
(now that I've remembered which address is subscribed to this list)
Does anyone with one of these phones have any sort of presence
working? I'm looking to monitor the DND state of the phones, if
nothing else. Setting the SIP-B bit enables SUBSCRIBE/NOTIFY, but
the dialog package is the
You need the Red modules to connect to PSTN.
Where are you located, We may be able to do a swap.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Luz Lopez
Sent: Tuesday, February 14, 2006 10:04 AM
To: asterisk-users@lists.digium.com
Subject:
[EMAIL PROTECTED] wrote:
Hi,
I've just compiled asterisk and get errors when trying to run it. Am I
right in thinking that if my digium card is not plugged into our ISDN
line then this is normal. If I replace zapata.conf with a blank zapata
file asterisk runs fine. The reason I ask is we
Group:
I have a customer that is running the following
Asterisk CVS-HEAD dated 2005-08-18
WhitBox Linux respin 2
mysql Ver 11.18 Distrib 3.23.58
Cisco 7960G
We are using the real-time drivers for sip and everything is working
great.
They have a few employees that use the phones from home on
Jimmy wrote:
Can Asterisk send RTP to a specific port number? For instance, I know I
can limit INCOMING RTP to certain ports in rtp.conf, but can I limit
OUTGOING RTP to a specific port (specifically port 5004 - I'm testing a
theory and need to be able to do this)
No. SIP/SDP negotiation
Perhaps I'm missing something here, but why not just have asterisk
dial all the phones regardless? No need to check what's available or
not, just dial all of them. If you don't want users on the phone to
hear a call-waiting beep, just make sure call-waiting is disabled.
Any phones that are able
Eventually I will learn to read the message twice before responding.
I see you are dialing all the phones first. So if just one is busy
asterisk won't dial any? That's odd. Now, I don't have any phones on
a zaptel card (just an X101P for incoming), but when dialing multiple
sip phones, it'll
hello,
I have one account i need using multiple sipura ata, for my account.
it's possible in asterisk.
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Interesting, I seem to have gotten more 'artifact' complaints after
switching to the svn trunk MG2 echo canceler, I just tested with the
echo canceler turned off completely, the artifacts are still there.
I haven't heard the loud buzzing sound since I stopped using the
aggressive canceler.
I've
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I normally don't write emails like this, but the last couple of days
have been very frustrating. We built a system for a customer with the
following hardware:
Asus Vintage (sis chipset) P4 2.8
512M Ram
LSI SATA Raid card (2x80g drives)
Digium
On 2/13/06, Mike Pollitt [EMAIL PROTECTED] wrote:
Hi Rob –Is it possible
to disable the onboard echo canceller so that one may try the software
cancellers instead?
I have the
TE110P and am experiencing the same bad echo problems that I can't seem
to effect by fiddling
Got it.. so, in this case, I am using 36 licenses, right??
Thank you very much
Dov
- Original Message -
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 14, 2006 1:25 PM
Subject:
Wai Wu wrote:
If Asterisk is in the public network, it will work. The problem is when
Asterisk is behind NAT and one of the client is also behind the same NAT.
No, it won't. If one of the clients is behind a NAT firewall, you cannot
tell that firewall to start accepting media directly from the
Yeh my advice is don't, or if you must make sure you clean your dialling
list first with the DNC list.
Yes I know that political messages don't need to obey the DNC list but
they should.
This is why I don't donate t most political campaigns.
Dean
-Original Message-
From: [EMAIL
Ok,
after 10+ minutes of mindnumbing hold music ;) no artifacts with only
one TDM card installed, going to test with two cards on their own IRQ in
a minute again..
Feb 14 09:59:31 [kernel] Zaptel Version: SVN-trunk-r941M Echo Canceller:
MG2
Feb 14 09:59:31 [kernel] ACPI: PCI Interrupt Link
Hall, Eric M. wrote:
Asterisk CVS-HEAD dated 2005-08-18
WhitBox Linux respin 2
mysql Ver 11.18 Distrib 3.23.58
Cisco 7960G
We are using the real-time drivers for sip and everything is working
great.
They have a few employees that use the phones from home on a RR or DSL
line.
The
I was gonna say use a queue of sorts, throw the devices into the queue
and tell it to ring all. I haven't played with it, but I would assume
that if a line's in use, it won't ring that person.
Aaron
Joseph Tanner wrote:
Perhaps I'm missing something here, but why not just have asterisk
dial
How much time and hardware do you have to do this?
VICIDIAL is capable of doing this(and we have done that with VICIDIAL
in the past) with hundreds of concurrent calls if your servers can
handle it, but it takes a few hours to setup and if you are only doing
this once it may not be worth it.
When agent tries to transfer a phone call (*2 - att transfer) he hangs up.
Why? When a phone call isn't from queue then att transfer works fine.
In features conf I have *1 for recording, *2 for att transfer and #1 for
blind. In queue blind transfer works. For disconnect I have #0.
I guess
Reli Loin wrote:
hello,
I have one account i need using multiple sipura ata, for my account.
it's possible in asterisk.
No. Generally you never need multiple devices to use the same account
information. This has been talked about in the archives.
Personally I use the MAC address of the
On Tuesday 14 February 2006 10:56, Sean Cook wrote:
we had originally purchased a tdm2400 with echo cancellation but
couldn't fit it in the chassis.
Spent 3 - 4 hours tracking down the source of the echo to no avail,
enabled mark2 echo cancel and aggressive echo cancel to get the system
to
I am From Nicaragua.
Regards
From: Alexander Lopez [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users]
I am From Nicaragua.
Regards
From: Alexander Lopez [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users]
I am From Nicaragua.
Regards
From: Alexander Lopez [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users]
Ron Senykoff [EMAIL PROTECTED] wrote:
I'm helping out with a political campaign and would like to use asterisk
to blast out about 200,000 calls with a short message from the candidate.
Can you tell me which party this is for, so I can ensure I never vote for
them?
--
PGP key ID E85DC776 -
Unless you use SIP ALG (Application Layer Gateway) like the module in netfilter to set the expectations? correct?
RegardsOn 2/14/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Wai Wu wrote:
If Asterisk is in the public network, it will work. The problem is when
Asterisk is behind NAT and one of
While I'm thinking about it, is anyone else out there using podget or
something similar to do news/weather playback?
It's a neat idea, and I'd like to showcase it as a feature that the old
Nortel just didn't even come close to doing...
Thanks,
Bob McDowell
Hi Klaus,
I have a problem with bristuff and analogue phones off FXO ports on a
Digium TDM400 card.
First I ran just the TDM400 cards with the phones on asterisk-1.2.4 and all OK.
Then I changed to bristuff-0.3.0-PRE-1l (your latest version ) and then I ran
into a problem,
The analogue
hi all,
I have a problem with @ 1.2.4 on debian kernel 2.6.8-2-386.:
-- Executing Dial(SIP/2003-bbae, zap/2/03460816149|30|t) in new stack
Feb 14 17:25:25 WARNING[4246]: channel.c:2535 ast_request: No channel
type registered for 'zap'
Feb 14 17:25:25 NOTICE[4246]: app_dial.c:1011
Voipjet should be fine however dont have all the calls
go out at once. test your system first and see how
many concurent calls you can have at once without
loosing voice quality. I would also reccomend getting
a dedicated server to do the calls as apposed to
buying the equipment if you are doing a
I have installed Asterisk and when I hangup the zap channel Asterisk show this message: Feb 13 17:45:49 WARNING[1748]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 4: Red AlarmFeb 13 17:45:49 WARNING[1748]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation
why not try www.asteriskhelpdesk.com
(No, I am not affiliated with them in any way)
--- Dean Collins [EMAIL PROTECTED] wrote:
Hi all,
I was just on the phone with a B2C company in
Australia who are looking
to integrate an Asterisk solution with their
Salesforce.com CRM
platform.
well,
My fulltime job is to add (i.e code if have to) some instant msging to our
asterisk pbx. I have no idea, where to start. i want contribute write code,
test etc. pls advice.
there is SIP extension method called MESSAGE which get IM done. Does
Asterisk support both SIMPLE and XMPP.
I'm using realtime caching. Here is my sip.conf file
[general]
callerid=unavailable
context=default
allowguest=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
nat=yes
canreinvite=no
rtcachefriends=yes
allow=ulaw
allow=g729
All other information about the sip clint is keep in the db
Thanks
Just wanted to also say this does not happen to all users behind a NAT
box on RR or DSL line just a few.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, February 14, 2006 11:19 AM
To: Asterisk Users Mailing List -
Moises Silva wrote:
Unless you use SIP ALG (Application Layer Gateway) like the module in
netfilter to set the expectations? correct?
If that exists on every NAT firewall in the path to both clients, yes.
___
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Rick Smith wrote:
Phil;
What link ?
Your question is a bit vauge, but here are some relevent urls:
Sprint CoS request form (a 2 pager, with some great links to a
guidelines doc and faq):
http://www.sprintlink.net/maint/cos_template.cgi
QoS:
http://www.voip-info.org/wiki/view/QoS
Phil
Ron Senykoff [EMAIL PROTECTED] wrote:
I'm helping out with a political campaign and would like to use asterisk
to blast out about 200,000 calls with a short message from the candidate.
Can you tell me which party this is for, so I can ensure I never vote for
them?
It's a basic GOTV (Get
Bob,
There are issues with AstTapi and Windows XP. I suggest you go to the
Asttapi bug list (on sourceforge.com) and check for patches etc., I have
submitted a couple of bug reports with no response to date. I am in the
process (with some success) of debugging the issue myself, and can send you
a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 14 February 2006 10:56, Sean Cook wrote:
we had originally purchased a tdm2400 with echo cancellation but
couldn't fit it in the chassis.
Spent 3 - 4 hours tracking down the source of the echo to no
avail, enabled mark2 echo cancel
Why not? If the client that's behind the NAT is able accept media directly from
Asterisk (which is in the public network), it should be able accept media
directly from another client. Otherwise, the whole STUN scheme is not going to
work.
-Original Message-
From: [EMAIL PROTECTED]
David Choo wrote:
Hi,
Consider doing rate limiting / bandwidth reservation. It worked heaps of
wonders for mine!
That's good to hear. BTW- Am I doing this right? Here are the relevent
chunks of my config on my router:
!
!
class-map Platinum
match access-group 101
!
!
policy-map
Apparently I didn't do the first test correctly where I said I made sure
I had only two cards, each on their own IRQ and still got artifacts.
When I repeated the test today, with both cards on their own IRQ, I got
no artifacts at all, after shuffeling the cards around for a bit I was
able to get
Because it's a new service without much traction and I need something
now and also because this will be a fairly large project and as such I
don't want to use a third party service to collect a 'fee'.
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
I'm helping out with a political campaign and would like to use
asterisk to blast out about 200,000 calls with a short
message from the candidate.
I can't speak for anyone else, but I'd find it very difficult ethically to
be involved in this, even assuming it's legal.
Check very carefully
Wilson Pickett wrote:
I just bought a new IAXy box and am only achieving one way calling.
Both iax.conf and the IAXy support ulaw and gsm. When I try to call,
Does the IAXy now support anything but ulaw or alaw? The original one didn't.
Dont know. All i know is that i had ulaw
Have you looked at Wildfire (was Jive Messenger) with the Asterisk-IM
plugin? It seems to work fairly well in my experience. I have it running
here at home and also on one client's network. The XMPP client they provide
(Spark) is a bit primitive, but something like Trillian also supports the
We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This
should get rid of static on the line (at least any static generated by our
half of the circuit), right?
I am a total virgin to ISDN. I understand that we need two BRI circuits to
provide four voice channels, and that the
Greetings to all,
Can anyone think of a reason that a Softphone would not be compatible
with the F.C.C's order for E911? If the user is able to update their
address when they move their laptop, etc.
___
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Hi, I have some questions :
If a client connected to a S.E.R. use as codecs
only G729, and I want to call and give him a message in gsm or wav format using
the manager API from asterisk server? This will work directly or it's necesary a
codec converter? My asterisk has the codec g729 as
On Tue, 14 Feb 2006, Ron Senykoff wrote:
I'm helping out with a political campaign and would like to use
asterisk to blast out about 200,000 calls with a short message from
the candidate.
there is a special place in hell reserved for people who do this.
my advice: don't.
-Dan
I've got several issues with AGI/FastAGI
1. When an AGI script sends a command to Asterisk via stdin, why does Asterisk
block and not return a result until the command is complete? Specifically, the
dial command. If I send a Dial command to Asterisk, I don't get a return result
until AFTER the
On Tue, 14 Feb 2006, Ron Senykoff wrote:
It's a basic GOTV (Get Out The Vote) drive, with just a short message
to encourage people to come out to the polls. It has nothing to do
with asking for any money, etc. Just a short message to people who
belong to the party from their candidate.
I just
Hi All,
I have a Digium card in my Asterisk server configured as pri_net and I
want to introduce latency on it in order to simulate PSTN conditions
and test some echo canceller hardware. Is it possible to purposefully
introduce latency and echo in a controlled fashion in order to do so?
On Feb 12, 2006, at 6:25 PM, Mike Pollitt wrote:
Hi Rob –
Is it possible to disable the onboard echo canceller so that one may
try the software cancellers instead?
I have the TE110P and am experiencing the same bad echo problems that
I can’t seem to effect by fiddling with the echo
Hello,
I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like
to be able to direct an inbound fax call into my TNT, have it answer
the fax and send the image file over to Asterisk, or some other
system to deliver to an e-mail address(s). I'm not sure if I need
Asterisk to
Hi
all,
At our customer site
i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware
1.0.1.9. When someone dials the customer, the receptionist picks up, and does an
attended transfer (the 'grandstream way')to a collegue. Most of the times
this goes ok, but sometimes,
Bob, why not do an adaptation on the [EMAIL PROTECTED] weather service?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bob
hey all, trying to get a zap extension to work I can dial out normally with
it, but if I try to access any of the features (i.e. *97 for voicemail) the zap
channel doesn't hear it, and all i get is dialtone. Is there a dialplan setting
or something to make the zap channels recognize keys like
I use Wildfire with the asterisk IM plugin, and it seems to work really
well. I'm running Trillian pro (Trillian basic does not support Jabber,
it's an added cost option ~$25), and our clients are running Exodus. I
do know GAIM doesn't show the status, only that the user is
unavailable.
Hello People,
I was wondering if you could take a look at this script,
exten = 505,1,dial(iax2/6311${EXTEN},t,25)
exten = 505,2,playback(pls-wait-connect-call)
exten = 505,3,set(NewCaller=${CALLERID(num)})
exten = 505,4,Set(CALLERID(num)=0${CALLERID(num)})
exten =
Is anyone using the SNOM 360 as a reception console with Asterisk? We
are trying to have the ability to view whether an extension is on or off
hook, or ringing with the Snom, which seems to work fine. The issue is
that we are having difficulty picking up calls and transferring.
Anyone have
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