Kevin P. Fleming wrote:
Erik wrote:
IMHO Asterisk B should change it SSRC to tell Asterisk A the RTP source has
changed (and fix this timestamp gap)?
That's an interesting question; since Asterisk is not actually a proxy,
in point of fact the SSRC has _not_ changed, since Asterisk B is
Assuming 'fencer' is the same thing as an 'electric fence', this is
called stray voltage. The farmer should want to fix this for reasons
other than the phone lines. He is probably getting excess voltage on
all sorts of things near his fence: water pipes, water tanks, his local
power, etc.
My needs are simple and clear. I have an office with 2 employees, a
sublet (independent business) and myself. I work p/t from the office
and p/t from home (telecommute). My wife works p/t from home
(telecommute exclusively). My other employee telecommutes from
350miles (we fly her in every 3-4
While the gurus are sleeping, I'll ask a few questions to get started...
I'm assuming this is using some kind of Voip, correct? If not, can you let us know what cards are involved? What kind of phone? Is it IP or analog?
But, to be honest, none of that should hinder performance. What
Hi All,
Voiceroute http://www.voiceroute.net has released a new version of the DRUID
Web-interface for Asterisk. There is a free trial edition available as well
as a live online demo both can be accessed through the website.
Tons of new features and extensive use of Ajax makes it a very easy
and
Using FTP to configure 501.
Gets past Running... App = sip.ld
and: Welcome! Processing configuration...
This may take a few seconds.
Then it displays:
Config file error
Error is 0x4020
and reboots continuously, repeating the above.
Anyone seen this before?
Is this a corrupt *.ld file?
An
Hey Everyone,
I have a fairly simple question I guess, though I currently don't know
how to solve it.
At the moment every Agent in a queue has 2 Phones, a PSTN and a SIP Phone.
How would I be able to limit calls to an agent, that if he has a caller
on one phone and another call is forwarded into
Hi group!
I have been using asterisk 1.2 for quite some time and now I need to go back on
asterisk 1.0 (because of oh323 channel driver). One thing that I can't remember
is does asterisk 1.0 support native music on hold? If it does, how can achieve
it, because it doesn't work the same way as
I have now:
exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
;exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
;exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
exten = _9011Z.,104,NoOp(${DIALSTATUS})
I configured two trunks for my outgoing calls:
It seems that the problem only occurs when there is a redirect:
-- Executing Dial(Zap/19-1, SIP/123|40|t) in new stack
-- Called 123
-- Got SIP response 302 Moved Temporarily back from 10.139.2.5
-- Now forwarding Zap/19-1 to 'Local/[EMAIL PROTECTED]' (thanks to
SIP/123-878b)
Hi,
I still cant dial out on Zap and I really have no clue why.
I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4
ports, 31 channels each and able to receive incoming calls and fax
perfectly.
I've done this in my dial plan.
exten = 111,1,Answer()
exten = 111,n,Ringing()
Hi everybody,
can I somehow find out when an agent has logged of?
I mean what time he/she logged off at?
I can see when an agent logged on,
but I'm missing when he/she last logged off.
Any hints are appreciated.
Thanks
Kristian
___
--Bandwidth and
On Tue, 2006-04-11 at 07:56 +0400, Jean-Michel Hiver wrote:
Joseph a écrit :
Is there a way somehow to implement Asterisk with Credit Card Processing
(IVR system)?
Yes, using AGI.
~google asterisk voip-info agi
Cheers,
Jean-Michel.
That is a positive new :-/
Any pointers to a
Dumpolid Exeplish wrote:
I am currently using * with MySql server version 3.23.58 by using the
sql driver compiled in the asterisk-addon tar file. Whenever i try to
update to a later version of MySql, the channel driver brakes and the
client is unable to connect to the newer verson of MySql.
On Monday 10 April 2006 19:20, Mojo with Horan Company, LLC wrote:
Try booting with apic off, I think it's noapic kernel option. Notice
this is APIC and not ACPI, which you referred to. Then get your
boards on different REAL irqs.
Please do not open your mouth to spout nonsense if you do
Use asterisk... not only is it more centralized. But, you also don't
have to rely on the ATA supporting it.
On 4/10/06, Mike Diehl (Encrypted email preferred) [EMAIL PROTECTED] wrote:
Hi all.
I'm in the process of configuring a phone system for my family and friends.
I'm wondering if I
Hi,
understand that the bandwidth utilized for each call is dependent on the
codec used, wonder if Asterisk can monitor the total bandwidth utilized
and restrict/reject new calls when the resource is insufficient to
support them reliably?
Regards
Andy Tan
--
Andy Tan
[EMAIL PROTECTED]
--
I guess there may be some legal implications, but that sounds like a good thing. I'd counter the legal implications with having a right to know who is calling,unless this is for an anonymous hotline.
On 4/10/06, Steve Totaro [EMAIL PROTECTED] wrote:
Peter J Dean wrote: I am not sure how this works
On Tue, 2006-04-11 at 09:26, Tomislav Parčina wrote:
Hi group!
I have been using asterisk 1.2 for quite some time and now I need to go back
on asterisk 1.0 (because of oh323 channel driver). One thing that I can't
remember is does asterisk 1.0 support native music on hold? If it does, how
Personally, I'd do it on the asterisk server. If not, things may get confusing if you add any other devices later.
On 4/10/06, Mike Diehl (Encrypted email preferred) [EMAIL PROTECTED] wrote:
Hi all.I'm in the process of configuring a phone system for my family and friends.I'm wondering if I should
Dear Group,
I'm working on a call recording solution and would like to have the ability to
initiate a 3 way call based on an incoming call.
One party will be an AGI that I have other will be an outbound call via a
second T1 interface.
Does anyone have a working configuration for an Asterisk
This isn't an asterisk question, more of a telco question. Hopefully
someone on this list involved with telco can answer it.
I have a client who is using asterisk with analog phone lines. There
is a bad clicking noise on the line. Click Click Click.
We have narrowed it down to
Rob Lith wrote:
Hi Terry
Could you outline more what existing set-up they have already - do they
want to use existing PBX's , if so which kind etc, and these Asterisk
or Ciscos as the voicemail application? you could also add on call
recording as a carrot.
Do they have analogue or
Yes contact me off list.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Monday, April 10, 2006 11:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] asterisk credit card processing
Is there
Mark Coccimiglio wrote:
Hey all,
It such a shame that BRI technology is such a flop in the USA. For a
small office such as mine it would be a great product. So her goes my
question What is a known asterisk working BRI card that will
operate in the USA. I need to weigh price/quality.
I CAN VERIFY via aa dozen PRI from XO that yes indeed provide
incoming callerID on PRI. It arrives shortly after the setup message.
Hence the wait(1) required to display it.
Now if you are referring to sending caller name across PSTN - that
does NOT work since the terminating switch will do
Mark Coccimiglio napisał(a):
Hey all,
It such a shame that BRI technology is such a flop in the USA. For a
small office such as mine it would be a great product.
Hello,
My opinion comes from Poland(EU). I use * with a 10 Euro (12 USD) 1
BRI (2b+d) card named Fritz! (the exclamation mark
As to original poster of the how-to page information, I apologize for my
sloppy proof-reading, or rather lack of. The word answer is mis-spelled
the mis-spelling was not intentional and as a result NOT required!!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
This is a serious problem!
I have brought up this issue in four previous attempts to get some feedback.
I find it hard to believe that no one else is having this same problem.
Apr 11 13:27:36 NOTICE[4446]: channel.c:1906 ast_read: Dropping incompatible
voice frame on Local/[EMAIL PROTECTED],2 of
Kerry Garrison wrote:
Has anyone got any information on bulk provisioning of Linksys
SPA-941/94s? There is an overview in the admin guide but it refers to a
different provisioning guide that I haven't found anywhere.
You will need to talk to the sales/marketing folks at Sipura/Linksys.
The
Hi, I have
problem with two my asterisk servers (connection by IAX). First is
working as master (external dynamic IP with dynamic DNS service).
Second is slave behind NAT. The second is registering on first one.
Then both servers can make calls to another one.
The problem is that sometimes master
It works as I described in
an earlier post. The page comes through, BUT the phone also plays dial
tone from the phone when it auto answers
Except for the dialtone, it works
Asterisk 1.2b1 and the last Sipura software version
Using the Page application in 1.2.?? provides the same result with
Hi,
I`ve copied an example directly from the wiki at
http://www.voip-info.org/wiki/view/Asterisk+tips+findme,
and it doesnt seem to work. I want the call to be bridge when the called
party presses 1, to be sent to voicemail if the called party press 9 and just go
to the next extension if
Understood on the proof reading.
The curious thing is it seems to work, EXCEPT when the 841 auto answers,
it also plays dial tone as if one had hit the off hook ( speaker ) button.
This phone has the last rev software, and I can't seem to locate where,
if any, place one can make a change in the
I'm not sure if it's the same problem but your error message likely the same.
after i additing pridialplan=local in the zapata.conf i'm able to make
outboundcalls (located in germany)
marcus
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag,
Unfortunately, Linksys is reserving the provisioning tools/info to their official resellers. The idea is that you pay your Linksys reseller to provision your phones (does not make ANY sense to me all). As a service provider, we should be able to buy these phones and have access to the bulk
Vikram Rangnekar wrote:
Feel free to try it out and send us any feedback you may have.
Vikram,
A few issues.
1). Requires to be run on the Asterisk server via Apache. On a
production machine, I try to keep the services to a minimal.
2). I take issue with a script that does a chmod
Jim Rice wrote:
Anyone seen this before?
I'm not sure about that exact error, but I get these systems if I stuff
up the XML in sip.cfg or phone1.cfg (or the specifc phone equivalents).
--
National Manager - Special Projects
Melbourne / Sydney / Canberra / Hobart / London /
2/340 Gore
While the gurus are sleeping, I'll ask a few questions to get started...
I'm assuming this is using some kind of Voip, correct? If not, can you
let
us know what cards are involved? What kind of phone? Is it IP or analog?
I use sip protocol with broadvoice.
But, to be honest, none of
Any futher suggestions?On 4/10/06, Michael Strelnikov [EMAIL PROTECTED] wrote:
But in this case you have to define two users on both sides. It is not most likely.
On 4/10/06, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I meant dialled extension, not originating
extension.
like :
exten =
We have written something, so the answer is yes.
regards,
Paul Hales
Technical Manager
AsteriskIT
Joseph [EMAIL PROTECTED] wrote:
Is there a way somehow to implement Asterisk with Credit Card Processing
(IVR system)?
--
#Joseph
___
I am currently using * with MySql server version 3.23.58 by using the sql driver compiled in the asterisk-addon tar file. Whenever i try to update to a later version of MySql, the channel driver brakes and the client is unable to connect to the newer verson of MySql. I have also tried updating the
I have a customer who is reporting very bad echo in only one or two of his
12 call queues.
The phones are cisco 7960 and 7940 running SIP6.3.
No problem with echo when an extension is dialed direct, internally or
externally.
Has anyone seen this before? And has any ideas about how to get rid of
Have a look at http://www.voip-info.org/wiki/view/sipura+mass+deployment
On Tue, 2006-04-11 at 06:05, Kerry Garrison wrote:
Has anyone got any information on bulk provisioning of Linksys
SPA-941/94s? There is an overview in the admin guide but it refers to
a different provisioning guide that I
On Mon, 2006-04-10 at 19:25 -0500, Eric ManxPower Wieling wrote:
Anton Krall wrote:
I will try that and see what happens...
This server is a supermicro one.. Anybody else had issues like this on
supermicro? Any hints on how to resolv them?
If I remember correctly, supermicro bios does
Some more info:
Just tried this on a server without using any NAT and no port
forwarding, no masquerading, and I still have the same problem. So there
goes that idea. I do not know what this VNAK error means.
By the way, I am using the latest version (1.2.6) of asterisk, have also
tried other
Hey
Everyone,
I just setup a way
for our receptionist to turn on the auto attendant mode via her phone. I
setup one of the indicators to dail an extension which runs a bunch of code to
turn on/off the night mode. Is there a way using the BLF to turn the
indicator on (solid red) when night
I'm rolling back from * version 1.2 to 1.0. Now I need to know what features
from 1.2 won't work on 1.0. So far I know this:
- features.conf doesn't work (does 1.0 have features.conf?)
- in extensions.conf I can't use
- as priority I can't use n
- Set(CALLERID(name)=)
-
Hello to all
I would like to know some opinions of people that are using billing
tools for Asterisk.
Can you please advise me in wich billing tool to I use?
Thanks
Joao Pereira
___
--Bandwidth and Colocation provided by Easynews.com --
Shouldn't be too difficult... perl has some great payment modules:
check out Business::OnlinePayment
http://search.cpan.org/author/MOCK/Business-OnlinePayment-StoredTransaction-0.01/lib/Business/OnlinePayment/StoredTransaction/Unstore.pm
modules on CPAN
Joseph wrote:
Is there a way somehow
When you find out what's causing it can you let me know as I have 1
system that gets this error and the telco tells me everything is fine
with their equipment.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pimjai
Wesnarat
Sent: 11 April
It looks to be as if you have your PRI D-channel defined as a voice
channel in your zapata.conf. For example in the UK channel 16 is the
d-channel so zapata.conf contains :-
switchtype = euroisdn
pridialplan=unknown
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31
On Tue,
I do have that line. I also have all my phones defined by IP address. But all providers are defined by names.On 4/10/06, Michiel van Baak
[EMAIL PROTECTED] wrote:On 22:14, Mon 10 Apr 06, Michael Strelnikov wrote:
Hi,My * refuses SIP registrations when internet is down. All is returing at the
Hi Marcus
Yesterday I tried that but it didn't work but today I tried again just
as u said and it works!!
Danke schön! Vielen Dank!
Gruß,
Pim
[EMAIL PROTECTED] wrote:
I'm not sure if it's the same problem but your error message likely the same.
after i additing pridialplan=local in
On Mon, 2006-04-10 at 21:42, Lonnie Abelbeck wrote:
Adding defaultip=10.x.x.x might solve the problem.
GXP-2000's can work without registering, using host=10.x.x.x as long as you
don't want to use BLF with the new firmware.
The new firmware is great, as long as you don't have early
Hi Tony,
thanks for your answer!
I tryed doing so, but I still get that error, sorry.
Giuseppe
-
Tony Mountifield ha scritto:
In article [EMAIL PROTECTED],
Giuseppe [EMAIL PROTECTED] wrote:
Hi,
when I try to use meetme I always hear this error message
"this is not a valid
maybe this could be solved using Local channel as members, and limiting calls to the agent (actually an extension if using Local/[EMAIL PROTECTED]) in the dialplan with GROUP and GROUP_COUNT
2006/4/10, Marco Campos [EMAIL PROTECTED]:
Instead of call-limit=1 try o use incominglimit=1. Note that
Hi everybody,
A customer requires G726-40 with Asterisk... I know G726-32 is
pseudo-standard, but he definitely wants G726-40...
Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone
done this before? Any hints? Please help!
Due to a misunderstanding, my product manager
Hi,
understand that the bandwidth utilized for each call is dependent on the
codec used, wonder if Asterisk can monitor the total bandwidth utilized
and restrict/reject new calls when the resource is insufficient to
support them reliably?
Regards
Andy Tan
--
Andy Tan
[EMAIL PROTECTED]
--
Out internet connection was out this morning. It seems that the SIP
extensions on our LAN were affected. Behavior like:
Call comes in over POTS to a TDM400P, there is a delay then before the Cisco
79[46]0's start to ring.
If we were lucky enough to get a call through, then we could not transfer
You could find here an xml example to provisioning them.
http://www.sipura.com/support/spa941faq/index.htm
Kerry Garrison escribió:
Has anyone got any information on bulk provisioning of Linksys
SPA-941/94s? There is an overview in the admin guide but it refers to
a different provisioning
On 4/11/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
Kevin P. Fleming wrote:
Ronald Wiplinger wrote:
It does not go to the next provider. Is there a settings for timeout
to go to the next provider???
Uhh... yeah. That is why there is a timeout parameter for the Dial()
if you have patched asterisk with bristuff, you could use the app DevState(newstate).
Basically, a thing like this:
; suppose 999 is your nightmode enable/disable extension
exten = 999,hint,DS/nightmode
exten = 999,1,your enable/disable stuffexten = 999,2,your enable/disable stuff
exten =
Just doing some test installs of asterisk running on branch (noticed first
on branch), and noticed if you move to virtual terminal 9 (may be
different on everyone else's), the CLI is running. Anyone have any idea
how to turn this off?
--
Aaron Daniel
Computer Systems Technician
Sam Houston
Just thought I would post this as someone might find it usefull.
This is the dialplan for making outbound calls from the UK (not
internetional).
It can be set to block callerID for particular extensions. I have also
added some detection of the PRI error numbers when a call fails to give
some extra
Leave it in the ATA! As long as you can live with standardizing on an
ATA. Sipura/Linksys a good choice for VSCs.
Even if you use some IP hardphones as well you will find softkey or
hardkey functions to replace most of the VSCs. (redial, forward, return
call, etc.). IP 501 Polycom has a good
because, a this time, the sip stack doesn't have asynchronous DNS... so ALL the sip code is stucked waiting timeouts for DNS queries (that are long timeouts).
When you try to dial a LAN device, the sip code is trying to resolve your voISP service providers' addresses.
We workaround this putting
Scratch that :) Figured it out.
On Tue, 11 Apr 2006, Aaron Daniel wrote:
Just doing some test installs of asterisk running on branch (noticed first on
branch), and noticed if you move to virtual terminal 9 (may be different on
everyone else's), the CLI is running. Anyone have any idea how to
Yes sir.. a fencer is an electric fence.We talked to Verizon
at one point and they really seemed to not care at all.
On 4/11/06, Bob McDowell [EMAIL PROTECTED] wrote:
Assuming 'fencer' is the same thing as an 'electric fence', this is
called stray voltage. The farmer should want to fix
BTW... another interesting issue.. the phone line (on Verizon's end)
run parrallel to the fence for a good 1/2 mile.
On 4/11/06, Matt [EMAIL PROTECTED] wrote:
Yes sir.. a fencer is an electric fence.We talked to Verizon
at one point and they really seemed to not care at all.
On 4/11/06,
+++ Doug Lytle [11/04/06 07:58 -0400]:
Vikram Rangnekar wrote:
Feel free to try it out and send us any feedback you may have.
Vikram,
A few issues.
1). Requires to be run on the Asterisk server via Apache. On a
production machine, I try to keep the services to a minimal.
2).
Hi Andrew...
Thank you very much for the info.
I didn't recompile the kernel, Im using a generic 2.6 kernel but its worth
taking a look at what you said.. Where can I find (which file) the Hz the
kernel was precompiled to?
Also, Im running 1 te110p and 2 tdm cards, probably I'll disable 1 card
Zttool shows no irqmisses on the te110p card?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Kenneth Lussier
|Sent: Tuesday, April 11, 2006 7:05 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] te110p
Out of the Box probably not but with an AGI script this is very
doable:
You can have a script that monitors active calls and the Codecs that are
in use. The script will have to do some math to calculate the bandwidth
in use and then using the variables in Asterisk, Namely SIP_CODEC. If
you are
On 4/11/06 8:42 AM, Carsten Bock [EMAIL PROTECTED] wrote:
Hi everybody,
A customer requires G726-40 with Asterisk... I know G726-32 is
pseudo-standard, but he definitely wants G726-40...
Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone
done this before? Any hints?
On 4/11/06 8:14 AM, Joao Pereira [EMAIL PROTECTED] wrote:
Hello to all
I would like to know some opinions of people that are using billing
tools for Asterisk.
Can you please advise me in wich billing tool to I use?
Thanks
Joao Pereira
___
Title: Re: [Asterisk-Users] Asterisk stops responding when internet is down
Asterisk is sensitive when it comes to DNS lookups. If the DNS server configured on your Asterisk server is not reachable, Asterisk may block while waiting for a result. This can cause chan_sip to hang and not allow
If the agent logs in as an agent thast is a member of the queue, then if
that agent is in multple queues they will only get one call at a time,
regardless or how many queues they are members of. (I hope you were able
to follow that!!)
In regards to the two phones.
1 WHY?!
2 DO
XO CAN supply callerid NAME on a NI2 PRI connection.
We have three of them and they work great. Its takes a little doing to
get to someone at XO that knows what they are doing
but XO does have some VERY good tech support people that know how to
get things done. It just takes a bit of work
On 4/11/06, Andy Tan [EMAIL PROTECTED] wrote:
Hi,
understand that the bandwidth utilized for each call is dependent on the
codec used, wonder if Asterisk can monitor the total bandwidth utilized
and restrict/reject new calls when the resource is insufficient to
support them reliably?
Hopefully it's okay to *announce* this here.
I've been working on a project for Asterisk for some time and it is
finally ready for a beta release. Any feedback is well appreciated. At
the basic core it's a Dialer for Windows. I'll be adding more features
quickly, but I wanted to keep everything
Hi everybody,
A customer requires G726-40 with Asterisk... I know G726-32 is
pseudo-standard, but he definitely wants G726-40...
Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone
done this before? Any hints? Please help!
Due to a misunderstanding, my product manager already
On 4/11/06, Rich Adamson [EMAIL PROTECTED] wrote:
In the US, bri pri's are less popular for lots of reasons, part of
which is the cost of implementing the necessary software on the CO
switch. Siemens (as one example only) charges their small CO customers
$7,000 to implement the software
Can anyone provide any further info on External IVR application? It
seems interesting. I currently have a heavily used AGI script that I
use for a custom IVR. It is written in Perl. I wonder if it would be
more efficient to migrate it to this External IVR. Will it be more
efficient? Will
No, I'm taking receiving CallerID name and *not* sending. and no on a
PRI wait should not be required for callerID to come in.
On 4/11/06, Jerry Jones [EMAIL PROTECTED] wrote:
I CAN VERIFY via aa dozen PRI from XO that yes indeed provide
incoming callerID on PRI. It arrives shortly after the
Hi,
Gere are some messages that sometimes show up in my
Asterisk logs... If you help me out to solve them, I could make a list of all
know warning messagesso that we can publish in the wiki or somewhere
else!
- "res_features.c: Did not read data." - on Google,
the only reference to this
I've had this problem too. It would get so bad, that it wouldn't even
answer incoming calls, and if I tried to dial out via pstn, I would
have hung up before it got around to dialing (which it would
eventually do, unfortunately).
A short-short term solution was to install bind, and use it as
Does anyone have a SEPMAC.cnf.xml file that works with asterisk? I
have the SIP firmware loaded on my Cisco 7970 but the status log shows
errors parsing the config. I copied a config that was posted to the list
but it didn't seem to work. Any help would be appreciated.
Jeremiah
--
Hi Anton,
I'm using a supermicro P4 3GHz P8SCT (Intel E7221 chipset) with TE205P and a
TDM04 and I've similar problem.
I was using linux 2.6.9smp that seams to have problem with APCI so
Hyperthreading, even if enabled, was not working (I sow 1 cpu).
Today I've disable hypertrading and start using
On Tue, 11 Apr 2006, Andrew Kohlsmith wrote:
Please do not open your mouth to spout nonsense if you do not know
what you're talking about.
[...]
Again, if the IO-APIC is reporting that the card is on its own IRQ,
it really, truly, honestly *IS* on its own IRQ. The
I have an * box that I need to chang the IP address on.
My hope was that I could add an alias to the interface with a different
IP address, have * bind to all addresses, change DNS and when no more
hits come on the old address.
However, IAX registrations coming in to the alias don't seem to get
Hi niklas,
I know it would have been
span=2,1,0,ccs,hdb3,crc4
But if I try this configuration asterisk zapata seams not to be able to
sincronize.
On logs I have continuosly:
Apr 7 09:03:18 NOTICE[3196] chan_zap.c: PRI got event: Alarm (4) on Primary
D-channel of span 2
Apr 7 09:03:18
Hi,
I've been battling with a similar issue:
a) I wrote a script to periodically run the command cat
/proc/interrupts and figure out the interrupts per second. I run this
script for over 24 hours and never once did the difference between the
preceeding and succeeding interrupt counts go below
Linksys just lost my VoIP business I guess.
-Original Message-
From: tracinet [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 11, 2006 8:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA-941/942 Bulk provisioning
Unfortunately,
Waldo Rubinstein wrote:
Can anyone provide any further info on External IVR application? It
seems interesting. I currently have a heavily used AGI script that I use
for a custom IVR. It is written in Perl. I wonder if it would be more
efficient to migrate it to this External IVR. Will it be
Nobody knows the answer to this!?!?!?
--
~Shaun
I have a macro that runs off a dial() and gives the callee a bunch of
options... one of them is to disconnect the caller. I read that setting
MACRO_RESULT=ABORT would hang up both legs of the call. When i set
MACRO_RESULT=ABORT and return to
Hi List,
I'm writing a system that issues a lot of automated calls, in an opt-in
basis. I've found that even though the calls are to different
destinations, if they are issued within the same second, they get the
same channel AND the same unique id.
Is there a way to prevent this?
That is a positive new :-/
Any pointers to a sample? I couldn't find a suitable sample. I don't
have much experience with AGI but I can follow a sample if I had one.
I usually call a bank's IVR and I'm asked for merchant number, device
number, etc. The system ask me for credit card number
Hi,
Do we need STUN server with Asterisk(1.2.6) for SIP phones which are using
NAT on different networks ???
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Erick Perez wrote:
How much RAM disk is needed or are you using for your current needs?
We're planning to do something like this. But I can't figure proper
dimensioning.
Erick,
We are using Asterisk to handle our inbound call center operations.
There are currently 158 leg files (produced
1 - 100 of 146 matches
Mail list logo
