I have trixbox installed but dont see a2billing installed together with it...anyone have integrate this before or is there any billing system that can integrate with freepbx. thanks
On 9/12/06, William Piper [EMAIL PROTECTED] wrote:
Both trixbox and asterisk2billing have their own lists... you may
[EMAIL PROTECTED] wrote:
Hi
I have a test setup of a sirrix card installed in NT mode connected to a
PBX. I keep getting the following error:
D-Channel receive message aborted, discarding frame (RSTAD=0x1c)
What does this mean? What could be causing it?
The answer comes a little bit
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi everyone,
I am looking to log CDR records to our MSSQL database for further
examination on the records. From what I gathered from the wiki I have to
choose between FreeTDS and unixODBC. Is there a better choice? Which
option
Hi,Are you aware of any SIP hardphone (or softphone) offering the option to browse its missed call list from a distant (xml, SQL or whatever) server instead of using its own list ?This would be very useful to avoid duplicate entries for instance, when an incoming call is forwarded from one
Hi Alan,
simply do not use Dell hardware.
we had your problem, we called Dell and they told us that our servers
was not configurable (they were too cheap).
So now I do not use Dell anymore and we have less problem.
Giorgio Incantalupo
Alan Bunch wrote:
I was going to use a Dell 1425 for
Hi,During off-hours, we often set Asterisk server to ring all extensions.Sometimes, some of these are diverted to mobile or off-site numbers.Which is the best way to handle this ie to make sure only not diverted extensions are ringed ?
My understanding is Asterisk cannot know in advance which
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
The loadInformation line in the SEP file reads like this:
loadInformationSIP70.8-0-4SR1S/loadInformation
- Here are TFTP server logs to illustrate that I'm using the correct
case'd XmlDefault.cnf.xml file:
Sep 10 21:57:55 bubbles
Hi all.
I've got a question regarding usages with a TDM400.
In results from the command
$ asterisk -rx 'zap show channels [n]'
I've noticed a change that may have been made somewhere between
ver 1.2.9.1 and 1.2.11.
In the older versions, the line Real: seemed to contain the text
'Linear' if
Hi all,
I am trying to register myself to my VOIP provider (budgetphone.nl) so I
can accept inbound calls. However, using sip debug I get the following
error:
-- SIP read from 81.23.228.150:5060:
SIP/2.0 403 Relaying denied
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK65bc02f0;rport=1048
Hi list,
I am searching for a possibility to let my * log per day. So that a new
logfile is taken every night at midnight, with the date in the file name.
Is there a way to do so? Does anyone of you has tried that before?
Regards, Christophorus
___
If you have four pstn telephone numbers (eg, 444-1212, 444-1213,
444-1214, and 444-1215) from your telco, then call the telco and have
them implement call forwarding on each of the four lines. You might also
verify they provide a call forwarding on busy function for those lines.
After they
You could use logrotate or you could configure your cron to send
asterisk -x logger rotate, which it will do what you want.
Regards
Christophorus Laube escribió:
Hi list,
I am searching for a possibility to let my * log per day. So that a new
logfile is taken every night at midnight, with
Kind time of day, All!
Prompt, please!
Is Samsung OfficeServ 500 in it card TEPRI is established, also there is
a server on Gentoo with Asterisk PBX + Established card Tormenta 2 (4
ports PRI). On Asterisk awakes it is submitted 3 PRI a stream from PSTN
and from it in turn on Samsung
Problema solved!
Just put resetinterval=never inside zapata.conf
Giorgio Incantalupo
Giorgio Incantalupo wrote:
Hi,
I installed an Asterisk box with a sangoma A102 PRI card. Sometimes
Asterisk drops calls...there is nothing inside logs but these warnings:
Sep 11 15:00:18 WARNING[3503]
Hi,Who has experienced using misdn instead of bristuff with Junghanns BRI cards inside a 1.2 Asterisk server ?What was it like ?Any advice about that ?Regards
___
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asterisk-users mailing list
To
On Mon, 2006-09-11 at 20:25 -0700, Christopher Corn wrote:
im having issues when routing calls from the outside with my new VSP.
this is what asterisk tells me when i try to make an incoming call, i
get the no service response when i call.
-- Executing GotoIf(SIP/christopher_corn-eddb,
For the curious, can anyone tell me how this flag fixes the issue? - I
have seen the error before, but always assumed it was related to hung
channels.
Thanks,
Steve
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Problema solved!
Just put resetinterval=never inside zapata.conf
well, not to disappoint anyone, but GrandStream Phones DO support remote provisioning. It's only a matter of setting it up. But with tftptext editor you can set every feature and update firmware remotely.You could also reboot the phone using CURL.
Anyway, i think in new GXP-2000 firmwares also
This is a questions about database verification and not a2billing. Asterisk also uses database for such things as cdr and sometimes you call dial plans from database. Someone might have seen a similar situation while installing postgres for Asterisk. It is Asterisk related.
--
Steve Davies wrote:
For the curious, can anyone tell me how this flag fixes the issue? - I
have seen the error before, but always assumed it was related to hung
channels.
Thanks,
Steve
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Problema solved!
Just put resetinterval=never
On 9/12/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Davies wrote:
For the curious, can anyone tell me how this flag fixes the issue? - I
have seen the error before, but always assumed it was related to hung
channels.
Thanks,
Steve
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED]
Thanks Jessee,
I've just sent an e-mail to Grandstream support asking if they are
planning in a near future to release a firmware implementing
alphanumeric callerid for Budgetone series.
When they answer me, I'll replay to this thread with their feedback, so
the community can also benefit...
Hi,RFC3611 provides a way to monitor call quality.Do you know any SIP hardphone implementing this feature ?I'm aware of softphones doing so (Counterpath's EyeBeam, for example) but no hardphone yet.
Cheers
___
--Bandwidth and Colocation provided by
Hi Olivier,
I used bristuff then I passed to misdn.
But there are pros and cons:
- misdn is easier to install and configure (I do not know if bristuff
installation has been improved...but it was a bit tricky when I used it..)
- misdn has its own config file (misdn.conf) and does not use zap
doing it the dialplan way:if you log in your agents via AgentCallBackLogin, you can set the CallBack extension to an extension managed by a macro, where the macro will do what you need:extensions.conf
[agents-exts]exten = 100,1,Macro(stdagent|SIP/100|...)exten =
Hi,
I have a client who wants a call center with 16 analog
FXO modules.
I offered him a solution with a 1U or 2U rack and
Digium TDM2400 card.
I know that there mother board compatability issue
with the Digium cards, so
can anyone suggest a barebone 1U or 2U server (I
prefer the SuperMicro
On 9/12/06, Antoine Megalla [EMAIL PROTECTED] wrote:
Hi,
I have a client who wants a call center with 16 analog
FXO modules.
I offered him a solution with a 1U or 2U rack and
Digium TDM2400 card.
I know that there mother board compatability issue
with the Digium cards, so
can anyone suggest a
Steve Davies wrote:
On 9/12/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Davies wrote:
For the curious, can anyone tell me how this flag fixes the issue? - I
have seen the error before, but always assumed it was related to hung
channels.
Thanks,
Steve
On 9/12/06, Giorgio Incantalupo
shadowym wrote:
I found that the distortion was consistent. In other words it happened in
the same way at the same time in a particular file. I suspect it has
something to do with how Asterisk plays it back and not any sort of
hardware/IDE/interrupt issue. Kris, the developer of Astlinux
BenjaminI've
already read all voip-info's articles. The address you've mentioned
shows how to configure a DYNAMIC RealTime, not a STATIC one.
I've tried to use the same table with both realtime modules, but it didn't work. No users have been found (sip show conf).
If you could help me to solve
Shawn Kelley wrote:
Hi,
Does anyone know how to do a re-map of a key on the Polycom to make it
dial a number.
I know how to remap a key to a certain function, but I don’t know how
to make it dial a number.
I’m wanting to re-map the “Service” key to dial *8 for a group pickup.
Any help
On 9/12/06, John Marvin [EMAIL PROTECTED] wrote:
shadowym wrote:
[snip]
Asterisk not padding files to
even 20ms increments when playing them. So, although that may be a bug
in Asterisk, I thought I would see if that was the problem by writing a
quick C program to pad all my ulaw files to
I tried a lot of SIP and IAX softphones looking for ones I liked, noticing
some have certain features and others did not. For things like call
transfer, call park, group pick-up, line presence, and all those kinds of
extras I have a bit of confusion on where it is implemented?
Are these
Hello NG,
We've a small problem using agents in asterisk. One requirement is, if there
no agent logged into a queue, it shouldn't be possible that a call joins a
queue. I can configure that using the parameter joinempty=strict in
queues.conf, unfortunately the parameter takes only effect if I add
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
I get many of these warnings inside Asterisk log:
WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel
0/1 already in use on span 1. Hanging up owner.
What does they mean??
Can I assume then that
I have this line in my queues.conf:
announce= support-department
and I have an recording file
support-department-recording.wav file.
Can anybody tell me how to setup support-department
so it play the .wav file when agent pickup the phone? Where should I define
support-department so asterisk
On 9/12/06, Steve Davies [EMAIL PROTECTED] wrote:
I don't suppose you know what the silence padding bytes would be for ALAW?
Found it... It is 0x55.
Thanks for the program :)
Steve
___
--Bandwidth and Colocation provided by Easynews.com --
I'm using a Dell SC1430 that includes the Intel NIC and don't have any
problems at all. Also using a TE210P and TDM400P w/ 4 FXS in the box. I've
never had to reboot the box or restart Asterisk (except for kernel upgrades
and * upgrades of course).
-Brodie
On Monday 11 September 2006 05:12
Hi, Alan,
We use Dell 1850 (about 20 server) and we have 4 ports PRI Digium cards
in it and it works perfect.
It is almost PlugPlay.
greetings
Arjan Kroon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giorgio
Incantalupo
Sent: dinsdag 12 september
Hi all,I'm developing an IVR that will have to make some MYSQL queries and diferent DTMF menus. Preventing already my development effort, future I plan to deploy my own website where users can build their own IVR.
Would you recomend me to make it with Realtime Extensions, do it directly in
Is there a way to contact her directly or do we have to go through
Digiums website?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid
Bender
Sent: Sunday, August 27, 2006 11:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
-Original Message-
From: Adam Goryachev [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 12, 2006 6:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Soundpoint Key Remap
Shawn Kelley wrote:
Hi,
Does anyone know how
Great! Much appreciated, I'll do some investigation myself, I'll be visiting
Grandstream this week.
Jessee J Holmes
-Original Message-
From: Ricardo Carvalho [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: 9/12/06 7:11
Hi,What would you suggest to implement directed call pickup on bristuffed Asterisk 1.2 ?I'm after tle ability to pick a specific ringing call (without caring about which call arrived first, for example).
Something like : *8 + local extension would be perfect.voip-info.org introduces many paths
Email her directly [EMAIL PROTECTED]
Don't forget to 'donate' the recordings back to Digium for inclusion.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Savoy, Kevin - Williston, ND
Sent: Tuesday, 12 September
Kristian Kielhofner wrote:
Hello everyone,
I am trying to build zaptel 1.2.9 for AstLinux. I have already done
an svn export of the 1.2.9 tag, so I am not experiencing the missing
octastic issue.
However, I am having a funny problem. The zaptel.log that I have
attached tells the
I've ever post this question many times on asterisk
users without success ?
My config :
SER = outbound proxy presence/im server
ASTERISK
||
||
proxy/SER ===sip agents
+
rtpproxy
If a sip agents dial local uri
The lcd in the current budgetone series cannot support alphnumeric display.
Craig
- Original Message -
From: Ricardo Carvalho [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, September 12, 2006 8:11 PM
On Tue, Sep 12, 2006 at 04:33:21PM +0200, [EMAIL PROTECTED] wrote:
I've ever post this question many times on asterisk
users without success ?
asterisk-dev is not 2-level support for asterisk-users . Please
follow-up there.
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755
Hi all,
we plan to install several IAX softphones.
http://www.voip-info.org/wiki-Asterisk+IAX+clients
lists a lot of IAX phones for Windows and Linux. Which
one would you recommand? We will install IAX client on
Linux and Windows.
thx richard
__
I am encountering an intermittent issue where some of my calls are being
dropped. Most of the calls that are made are successful. However, some
calls will be dropped after having been connected for some time.
Each time a call gets dropped, I get output similar to the following in
the Asterisk
Hello,
I am trying to set conference system that will allow to bridge pstn and
voip conferences together,
So far I did this
created meetme conference room
conf = 500|1234
I created test extension 555, which does this:
exten = 555,1,MeetMeCount(500|count)
exten = 555,2,Gotoif,$[${count} =
Thanks for your answer.Has anyone followed the other way (msidn
on Junghanns board), as this would certainly prevent Junghanns nor
anyone else to provide any kind of support.Cheers
___
--Bandwidth and Colocation provided by Easynews.com --
On 9/12/06, Steve Davies [EMAIL PROTECTED] wrote:
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
I get many of these warnings inside Asterisk log:
WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel
0/1 already in use on span 1. Hanging up owner.
What
On 9/12/06, Olivier [EMAIL PROTECTED] wrote:
Hi,
What would you suggest to implement directed call pickup on bristuffed
Asterisk 1.2 ?
I'm after tle ability to pick a specific ringing call (without caring about
which call arrived first, for example).
Something like : *8 + local extension
I have a 7970G with 5.0.3.0S Skinny (Load File: TERM70.5-0-3-0S) on
it and I'd like to get it up to 8.x.
- With the SEPMAC.cnf.xml in place (which was taken from voip-info
(http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP
under This worked for me...)), I get Load ID
Hi,
thanks to all
I solved the calls dropped problem, it was resetinterval parameter in
zapata.now asterisk does not drop calls anymore.
I do not get the message:
WARNING[3503] chan_zap.c: Got restart ack on channel 0/6 span 1 with owner
anymore...but I get all the others.
I'm interested
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Antoine Megalla wrote:
Hi,
I have a client who wants a call center with 16 analog
FXO modules.
I offered him a solution with a 1U or 2U rack and
Digium TDM2400 card.
I know that there mother board compatability issue
with the Digium cards, so
Alan Bunch wrote:
I was going to use a Dell 1425 for Asterisk build but I see on Digium's
website that hardware may be problematic. Can anyone shed a litle more
light on the problem. I see the Intel ethernet cards seem to cause
problems. If I need to disable the onboard Intel on the Dell
Kohler, Jeffrey wrote:
I am encountering an intermittent issue where some of my calls are being
dropped. Most of the calls that are made are successful. However, some
calls will be dropped after having been connected for some time.
Each time a call gets dropped, I get output similar to the
On Sep 12, 2006, at 7:32 AM, Antoine Megalla wrote:
Hi,
I have a client who wants a call center with 16 analog
FXO modules.
I offered him a solution with a 1U or 2U rack and
Digium TDM2400 card.
I know that there mother board compatability issue
with the Digium cards, so
can anyone suggest a
Grandstream support just answered me saying that:
BT100/200 LCD does not supports alphanumeric caller ID display. You may
want to try GXP-2000..
It's confirmed! Future firmwares won't support that feature! :(
Thanks to all that replied,
Regards,
Ricardo.
Craig Guy wrote:
The lcd in the
o o wrote:
Thomas,
Thanks for your help so far. I finally figured out
where 'debug level 10' dumps to. In reading the logs
there, it's telling me I'm out of licenses. I'm not a
math wizard by any means, but I would assume with g729
on the GXP-2000 and on the IAX trunk, I would only
Hi Arjan,
We're thinking about purchasing a Dell 1850 for a new production Asterisk,
could you detail your spec, ie processor, memory, raid or whatever, it could
really help me. We too have a 4 port Digium PRI, a TE405 and also a TDM22b.
I know our requirements could be different from yours,
On Tue, 2006-09-12 at 09:00 -0600, [EMAIL PROTECTED] wrote:
I've ever post this question many times on asterisk
users without success ?
As I and many others have probably noted.
I found then neatly filed in junk mail.
Perhaps you're getting your just deserts.
--
Dave Cotton [EMAIL
At 06:43 AM 9/12/2006, you wrote:
It's not a great answer, but since it's only a problem adding you
might just have to validate the codes the agents type in.
exten = _*8XXX,1,Answer
exten = _*8XXX,n,gotoif($[${EXTEN:1} 8032]?GoodOne)
exten = _*8XXX,n,goto(hangup)
exten =
Has anyone ever gotten the Polycom MyStat soft-key to do anything?
Setting the status to something like 'Away', does not generate any outgoing SIP
traffic from the phone. Calling into the phone either from a watched buddy, or
other number, acts as if the status was never changed. A call to
Title: Verizon ISDN service in NY Hunt Groups
Is anybody on here using PRI or BRI service in New York state with the trunks in a hunt group from Verizon??
I'm trying to setup a system and I've spoken to three people at verizon who all claim they cant put BRI or PRI circuits into a hunt
Hi Users,Im looking for recommendations on softwares for calling card implementation and post paid billing services. Please give some recomendatons based on your experiences. A detail of the pros or cons (if any) you faced with it would be highly welcome.
thanks in advance.Dan
Okay. I'm setting up my first Asterisk box and the only thing I want to do right now is get my Ekiga softphone to register with it. Here is how I have my sip.conf set up:-sip.conf-[general]context=default
Hi all,Let me add to my query.I would prefer to have an Asterisk GUI that has billing calling card solution all in one.Got any suggestions.?Thanks
On 12/09/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi Users,Im looking for recommendations on softwares for calling card implementation and post
test
___
Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet
!
Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos
expériences.
I've been playing a lot with AEL and one thing seems to be perplexing me,
it's regarding actions taken using the IFTIME command, and they don't seem
to be having much affect.
Here's my AEL script:
// Main menu configuration
context mainmenu {
includes {
default;
Bernie Courtney wrote:
I'm trying to setup a system and I've spoken to three people at
verizon who all claim they cant put BRI or PRI circuits into a hunt
group, I find that EXTREMELY hard to believe.
PRIs don't use hunt groups (Just found this out myself). An inbound
phone number will
Hi list
I'm a bit stuck connecting my fonebridge from redfone to my telco. I
have a E1 line (30 channels).
The configuration is sent to the fonebridge and the led lights up red
and stays that way. I am using CentOs 4.4 and zaptel . 1.2.9.1 (had the
same problem with CenOs 4.3 zaptel 1.2.6. Also
On a similar note, there is a get together 6-8pm on Wednesday evening in
Room 211, it's open to anyone involved with Asterisk, if you have any
questions Carl Ford is the contact.
I'm just reposting as I haven't seen many emails about this get together
and wanted to make sure everyone knew.
I
2006/9/12, Steve Davies [EMAIL PROTECTED]:
As far as I know, bristuff includes directed call pickup already,using *8ext. Read their ChangeLog I think it has notes on how to useit. If not, I am sure the list archives will have all the requiredinformation.
Cheers,SteveReading
Im told by Adam below that I can use a Speed Dial to accomplish this.However, I dont know how to map a speed dial to the key.I know the syntax for mapping a function to it ( IP_500 key.IP_500.31.function.prim=BuddyStatus/ )However, I dont know how to do a speed dial.Any one out there
Hi I have 5 telular mod SX5e https://www.telular.com/v2/html/products/product_display.asp?productID=94 is a great stuff, but i have a extrange problem with asterisk. Sometimes the sound is ugly or choppy. The telular alone work fine all the time
For example if i made 5 calls from asterisk to gsm
Hi Jason
loadInformation6 model=IP Phone 7970SIP70.8-0-4SR1S/loadInformation6
1. Stick with the 8.0.2 SIP image as it works best with asterisk...
at least for me (o;
- Here are TFTP server logs to illustrate that I'm using the correct
case'd XmlDefault.cnf.xml file:
Sep 10 21:57:55
Hi Shawn -
Unfortunately, on a Polycom, you can no longer remap a speed dial to a
key. You can set extra line appearances to be speed dials (I can show
you that, if you want), but none of the other keys. This feature used
to be available, but was quietly removed as of 1.5.x. If you want to
I note that the SC420 is listed as incompatible but the SC430
appears to be a slightly different beast in terms of chipset, the 430
has the newer E7230 as opposed to the E7221 - does this make a
difference to compatibility?
I have an SC420 in one office that works quite well. I think the
I have a TDM2402E card calling out to NuFone using IAX2
and after couple of minutes (6 min to be exact) the recipient
can no longer hear the caller.
The caller can continue to hear the recipient clearly.
After the 6 min I continued to listen to the call and I a could
hear the other person but
Hi,
We have experience problems with calls between MGCP ATA's and SIP
ATA's (Linksys PAP2-NA).
A call from MGCP or SIP to the other connects normally and the
conversation can usually last around 30 seconds and it becomes one-way
audio.
What I don't understand is how the calls can be set up and
Hello,
announce = support-department
plays support-department.wav so playing support-department-recording.wav needs
announce = support-department-recording
bye,
Zsolt
On 9/12/06, gc [EMAIL PROTECTED] wrote:
I have this line in my queues.conf:
announce= support-department
and I have an
The docs for 1.6.6, 1.6.7 and 2.0.1 say you can do it, and I did it yesterday
on 2.0.1
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 12, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Hello:
I'm would like to get feedback before finalizing design of a VOIP
network, in particular about people's experience with network (primarily
10/100/1000 twisted pair) ethernet switches.
I have a number of candidates in mind, but I would like any and all
opinions and suggestions on the
im finding companies like voxee that offer very low rates and then companies like voipstreet that offer at a higher rate doube. whats the catch? is voxee, what you would call a wholesaler and voipstreet, commercial?im worried about going with companies like voxeee, because i question their
Did you ever try to get it working on any 1.6.x releases? I hacked at
it a bit and it didn't seem to be working, though I could have been
doing something wrong. I was, after all, reading the manual... ;)
I'm glad to hear someone successfully doing it, as it's something I've
wanted to play with
You're right Voxee support sucks. But I think they do well and provide good rates. I'm using Gafachi, a little expensive and have Voxee. I'm using LCR so the termination will try Voxee first and when not available will use Gafachi. You can set up something like that with a least cost routing.
Hi Doug -
AFAIK, you will need to tell it save a speed dial for *8,
and then map
the key to dial the speed dial number that you saved it as.
The docs for 1.6.6, 1.6.7 and 2.0.1 say you can do it, and I did it
yesterday on 2.0.1
I just noticed that version 2.01 came out. I'm really glad
At some point in my dial plan, I need to find out the length of a sound
file in seconds (to weed out things that are way too short)
the record application doesn't seem to have any facilities to do that.
any ideas ?
___
--Bandwidth and Colocation
"All
circuits are busy now" makes perfect sense in my
PRI trunk is full.
How do I stop
asterisk from playing this recording when it is a wrong/bad
number?
I gat a call
today that a user was trying "all day" to call a number in Mexico and kept
getting the above recording.
I said, try
Even though it might work, you should realy consider using Quad span
T1s with channel banks, or Xorcoms Astribank solutions.
On 9/12/06, Matthew Fredrickson [EMAIL PROTECTED] wrote:
On Sep 12, 2006, at 7:32 AM, Antoine Megalla wrote:
Hi,
I have a client who wants a call center with 16
BerkHolz, Steven wrote:
All circuits are busy now makes perfect sense in my PRI trunk is full.
How do I stop asterisk from playing this recording when it is a
wrong/bad number?
I gat a call today that a user was trying all day to call a number in
Mexico and kept getting the above
Dear friends,
I'm trying to dial out using a INX (internationalnumber.com) line but I get the message "ths account number is not valid".My asterisk is working well with other providers. INX support told me the line is working and in fact when I setup this line on my softphone it works.
I was
Hi,
has anybody experience running asterisk on a (i.e. fedora-based) Xen
system? What about mISDN support etc.?
For a low-load system I thought about using:
1. Sempron 2800+
2. some memory, in your opinion how much should I attribute to the
asterisk guest system?
3. A AVM Fritz!PCI card for
I need h.264 and tried therefore svn checkout
http://svn.digium.com/svn/asterisk/trunk asterisk
(currently I have branches 1.2 installed)
make clean; make update; make install
.
make[1]: Entering directory `/usr/local/src/svn-versions/asterisk'
rm -f .depend
rm -f .depend
rm -f .depend
Hi, what mean this voice message that asterisk say when I try to call an
extension of another asterisk connected by IAX2 trunk?
This problem exist only if I call from asterisk1 to asterisk2, vice
versa all work.
___
--Bandwidth and Colocation provided
At some point in my dial plan, I need to find out the length of a sound
file in seconds (to weed out things that are way too short)
the record application doesn't seem to have any facilities to do that.
any ideas ?
use sox beep.wav -e stat and parse the output
man is your friend
google also
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