My experience with the TenorAX you could use H323 or SIP. And the only
supported codecs were G729 by default and could either be changed to ulaw or
alaw.
Thanks,
Steve Totaro
-Original Message-
From: Jonathan k. Creasy [mailto:[EMAIL PROTECTED]
Sent: Fri 3/31
if you reboot your phones from the asterisk server ie via cron or so,
that reboot script could potentially delete the phone-specific directory
xml before the sip message is sent
Noah Miller wrote:
Hi Avi -
I know this is off-topic for Asterisk, but I don't know where else to
ask: I've setup
Title: voicemail to email sending problems
I have a box that will send to my personal pop/web based email but will not send to my exchange server. I have checked the MX record and DNS settings. I know there is something you can do like this to check it but it returns either a -1 or 0 (have n
Guyz!
Is there any known issue with ooh323 and g729? I am experiencing one side voice okay from ooh323 extension to sip ext, but on reverse side voice quality is very poor. Any thoughts?
thanks in advance.
___
--Bandwidth and Colocation provided b
I would not ride on a tracert too much. We use Teliax also and our ISP that
we have at the data center switched there backbones around the same time
Teliax where doing there upgrades.
We started seeing some call issues and when we did a tracert we started
getting some dropped tracert responses on
Title: asterisk-stat and webmeetme by areski
I like to think I am not a complete idiot...
...I have googled till my fingers bled.
I cannot figure out how to install these apps.
I have figured out the database protion as well as editing defines.php but the web portion is killing me. I am runn
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
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> Something I've been curious about is if it is possible to stick their
> ata on a extra ethernet port on an Asterisk server and have the Asterisk
> server spoof the Vonage server. Then, do a man-in-the-middle type thing
> to use the ata for authentication, but have Asterisk handle all the calls.
[EMAIL PROTECTED] wrote:
it says that there's no packet loss and the average latency to teliax is
1/3 that of junction networks.
the traceroutes did on several occasions suggest that teliax's upstream
provider in colorado (rockynet) was suffering from bandwidth overload,
as the last hop from
On 3/31/06, Matt <[EMAIL PROTECTED]> wrote:
> Very good that's actually what I ended up doing.
>
> I think my confusion came up with "Hot Desk" use ... like where you
> want the agents EXTENSION to roam with them.. and agent-IDs.. which is
> what I wanted... once I figured out the difference things
I had this same problem with SX2000. I think you have to configure the Mitel to "Speed Dial" to your Asterisk server. What I think is happening is that the PBX grabs a trunk but does not "dial" into Asterisk. Ask the person configuring the Mitel PBX to setup an extension (e.g. 1234) when caller
I am installing one asterisk, to establish connection with my PABX Siemens, in ISDN, link went up normally, also I obtain to internally call the branches the PABX, normally, but when I try to dial for the PSTN, through pabx with the command exten = _ 19, 1, dial(zap/g2/${EX
it says that there's no packet loss and the average latency to teliax is
1/3 that of junction networks.
the traceroutes did on several occasions suggest that teliax's upstream
provider in colorado (rockynet) was suffering from bandwidth overload, as
the last hop from rockynet to teliax would j
> NICE!
>
> On 3/30/06, Joe Dennick <[EMAIL PROTECTED]> wrote:
> > I see (and like) the demo, but where can we get it?
> >
> > Doug Lytle wrote:
> >
> > > Nicolás Gudiño wrote:
> > >
> > >> Something like this perhaps?
> > >>
> > >> http://www.asternic.org/stats/demo
It is
Perhaps you would be willing to share it with the rest of the community?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender
Sent: Friday, March 31, 2006 7:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Us
mustardman29 wrote:
Not a lot to go on sam. What do you want to do? If you just want to play
or have very minimal requirements then get a soekris NET4801 board, CF and
install Astlinux.
http://www.soekris.com/net4801.htm
-Original Message-
From: sam [mailto:[EMAIL PROTECTED]
Sen
We are attempting to interface Asterisk with a Mitel 3300.
Initially we tried FXO Ground start T1 and had many problems with that.
Even spend 5 hours on the phone with a Digium engineer and were unable to
get it working right.
Next we turned it up as an E&M T1 and that actually worked great. No
p
> Appreciate the replies everyone -- really
>
> I'm wondering if I should be using zapHFC with my Junghanns card
instead
> of
> qozap? Everyone always mentions zaphfc -- mostly I guessed because
they
> are
> using a zaphfc-compatible card - but *maybe* I should try that
instead
> of qozap???
That was how I reset the black Iaxy I have used; I've never used a blue one.
What I found was the initial provisioning would work fine, but if I
tried to change the settings after having already provisioned the
device, the provisioning program would hang, so I Googled for
instructions on resetting
I Guess you can edit the following line in your hylafax config file for
your iaxmodems.
Class1RMQueryCmd: "!24,48,72,96"
Put exclamation in front of 96 (as it is done with 24) and it should
disable the receive with that speed.
>
> Is there a way to limit the speed of Hylafax to 7200
As a rule you are wasting your time trying to send calling name to
your telco. Unless your carrier is also the terminating providor for
everyone you call it will accomplish nothing.
Caller ID that the called party receives is by way of a lookup by the
terminating providor in a national data
Sent you a email
~Shaun
"Tom Vile" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
I have a script that will do this. Contact me off list for information.
On 3/30/06, Shaun <[EMAIL PROTECTED]> wrote:
> I am attempting to setup a asterisk server to take place of my current
> ser
Wynne, AR USA would be my guess.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andrew Kirch
> Sent: Friday, March 31, 2006 3:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] aster
I just installed Hylafax with Iaxmodem and I am not getting good
results when receiving faxes. I can see that the modem is reporting the
following:
Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 2400 bit/s
Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 4800 bit/s
Mar
> Flash Operator Panel already has similar functionality, just create a
> CID entry drag and drop. There may of course be other (better) ways to
> do this but this is one option/alternative.
You can also use FOP and javascript to initiate a call to the number
entered on a text input box, an href
> >
> And isn't mpg123 ( or some replacement ) required when using a stream
> for MOH
> I couldn't get streaming to work without it in 1.2?
Yes.. mpg123 is required for streaming... I had it working in 1.0.9...
though have not tried in 1.2.
___
--Bandwid
Very good that's actually what I ended up doing.
I think my confusion came up with "Hot Desk" use ... like where you
want the agents EXTENSION to roam with them.. and agent-IDs.. which is
what I wanted... once I figured out the difference things have been
going smoothly.. While I'm a long time as
Matt wrote:
Is it true that asterisk 1.2.6 does not use mpg123?
I just installed asterisk 1.2.6 and while I do have music on hold
(through format_mp3?) I do not have an mpg123 process running.
I seem to be having serious audio issues when going through one of my
providers (and just through t
So, if you are absolutely sure that you've specified the correct T1
synchronization parameters in your /etc/zaptel.conf and you still have
fax reliability issues, look elsewhere in your implementation for the
root cause.
So, would you conclude that it's possible for a given T1/E1
Is it true that asterisk 1.2.6 does not use mpg123?
I just installed asterisk 1.2.6 and while I do have music on hold
(through format_mp3?) I do not have an mpg123 process running.
I seem to be having serious audio issues when going through one of my
providers (and just through that provider) whe
On 31 Mar 2006, at 05:46, Leo Burd wrote:
Hello there,
I'm writing an application to display asterisk voicemail on a
webpage. Since Flash only handles MP3 files, I wonder if it is
possible to configure
asterisk voicemail in such a way that it would record and play MP3
files...
Would an
And, what does traceroute say about your connection with Teliax? Hmm?
[EMAIL PROTECTED] wrote:
On Fri, 31 Mar 2006, Michael Welter wrote:
Having said all that, I see where Teliax have installed the voip-co4
host on Viawest. Are you using that host for your analysis?
I have used every single
On Fri, 31 Mar 2006, Michael Welter wrote:
Having said all that, I see where Teliax have installed the voip-co4 host on
Viawest. Are you using that host for your analysis?
I have used every single gateway teliax has made available to me,
including their beta test ones. I experienced choppines
John Novack wrote:
One wonders why the original supplier isn't on the scene.
Was he in over his head?
Has the company failed to completely pay the original vendor, and is
attempting an end run?
Tread lightly until the complete picture is known.
My thoughts exactly.
Doug
--
Ben Franklin qu
I have an NI2 PRI connection to a 5ESS switch. I am explicitly told to
use NI2 as the protocol, not 5ESS.
I am running Asterisk 1.2.4.
When making calls to the PSTN, Asterisk sends the calling name as part
of the display information element in the SETUP message. I need to be
able to send the cal
Giuseppe wrote:
Can anybody tell me if there is some error or something missing in this
configuration please?
I have the same card in a few of my servers and the echo canceller works
just fine. I'm not 100% sure, but something does jump out at me:
== ISDN3: Answering for 'x'
-- Pl
Roughly where are you located?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of mike webb
> Sent: Friday, March 31, 2006 4:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] asterisk turn ke
>From voip-info.org
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf
[applicationmap]
; don't use e.g #9 for applicationmap or featuremap unless you have
changed 'blindxfer' from # to e.g. #1 !
testfeature => *9,callee,Playback,tt-monkeys ;Play tt-monkes to
callee if
One wonders why the original
supplier isn't on the scene.
Was he in over his head?
Has the company failed to completely pay the original vendor, and is
attempting an end run?
Tread lightly until the complete picture is known.
John Novack
Cory Andrews wrote:
Just
Hi,
For thanks to everyone that answered the "dial from pph".
On an other subject, how would I go about playing a wav file while
talking to someone over a Zap channel ?
Let me explain. I am on line with someone. I want him to hear a WAV
(or mp3) sound file. I punch a key on my phone ke
In futzing with my contact directory on my polycom 501s, I realized I
could put '[EMAIL PROTECTED]' to call the other phone --
without even touching the asterisk server. I suspect my phone (and the
remote phone) don't even have to be registered to an asterisk server, as
the username@ part does
Kevin P. Fleming wrote:
Michael Wallette wrote:
My only gripe is the initial configuration, although even that isn't too
terribly bad. You must download and unpack a C program, then edit a config file
that the C program pushes to the Iaxy. If you want to change
You can do provisio
hi Mike,
from where are u?
i know an Israeli company but the question is what capacity you need, where are you located, etc.
i may recomend you XorCom but as for the 24x7 support, i am not sure they are capable to provide you this.
Rgrds,
Mickey
On 3/31/06, mike webb <[EMAIL PROTECTED]> wrote
I agree we have this working also.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Boris Bakchiev
> Sent: Friday, March 31, 2006 8:55 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] As
You have to use H323 the last time I did
anything with their equipment. It has been almost a year but I think it went
fairly smoothly. Do you have a specific question?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen Arulraj
Sent: Friday, March 31, 2006 5
AAH 2.8beta1 include the FreePBX.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Thursday, March 30, 2006 9:44 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: FreePBX & AAH
In article <[EMAIL PROTECTED]>,
Jim
Hi Matt,
We have somewhat of a similar setup here in my office. We have multiple
queues to which different agents are a member to anyone of them.
Basically what I chose to do was make my own custom log in script. I
reference to the voicemail box and use the ID and password to
authenticate our
I've been quiet on this discussion for a few days and reading
everybody's thoughts. But since I brought this subject up in the
first place I thought I would bring it to a close with the results
summarized.
1. The polycom phones do support attendant stations, but there is
some incompatibi
Mike, here is the interrupts (sorry for the formatting)
CPU0 CPU1 CPU2 CPU3
0:117 174878327 0 0IO-APIC-edge timer
1: 0928 0 0IO-APIC-edge i8042
8: 0 1 0 0I
HiSomebody knows a hosted billing voip service ?roberto-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID:
[EMAIL PROTECTED]For reliable and professional DNS, use DNS Made Easy!http://www.dnsmadeeasy.com/u/14989
_
For 3804 default setting. Each 3804 fxo port can register to asterisk
server. And you can dial each extension and hear the second dial-tone.
But you can config the 3804 to one-stage dialing and just pass the phone
number to 3804. The 3804 will callout directally.
Regards,
Kevin
-Original Mes
Michael Wallette wrote:
> My only gripe is the initial configuration, although even that isn't too
> terribly bad. You must download and unpack a C program, then edit a
> config file that the C program pushes to the Iaxy. If you want to change
You can do provisioning from within Asterisk, after e
ADEGOKE ARUNA wrote:
> I didn't want to bother the list too much. However, after reading I discover
> I don’t have a clear cut way of doing transcoding.
You have posted this three times to the ss7 list and now twice here...
we get it, you don't know how to read the information on the wiki about
i
Dov Bigio wrote:
> Mar 31 12:53:04 WARNING[17170]: app_queue.c:519 statechange_queue: Failed to
> create update thread!
This means your system is out of resources (memory or something). The
state change will still be handled, but it will happen in the calling
thread, which increases the chances
MeetAsterisk - a one day training arranged by Digium, Voop and
Edvina.net - is arranged in seven European cities
during April and May. Learn more about Asterisk, test Asterisk
equipment and meet Asterisk professionals!
Go to http://www.meetasterisk.com to learn more and register today!
Regar
On Monday 27 March 2006 20:17, Benoit Panizzon wrote:
> On Friday 24 March 2006 16:05, Benoit Panizzon wrote:
> > Hi all
> >
> > Apparently there is a patch for those 1.2.4/5 MeetMe Freezes:
> >
> > http://bugs.digium.com/view.php?id=5884
> >
> > Haven't tried it out yet.
>
> I can now confirm: No
On Fri, 31 Mar 2006, Bob McDowell wrote:
> I agree, this does work well. My 'fax' extension is right off of the
> docs:
>
> -
>
> [faxin]
> exten => fax,1,UserEvent(Incoming Fax...)
> exten => fax,n,Dial(IAX2/ttyIAX)
> exten => fax,n,Dial(IAX2/ttyIAX2)
> exten => fax,n,Dial(IAX2/ttyIAX3)
> e
I'd be very interested to know how your Audiocodes install goes.
My experience was not good at all with an MP-108. It was very inconsistent,
and extremely hard to configure. I paid $275 for support from ABP, the
Audiocodes USA support provider, which was a waste. Thier answer was mostly
"oh
Flash Operator Panel already has similar functionality, just create a
CID entry drag and drop. There may of course be other (better) ways to
do this but this is one option/alternative.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of
use the manager interface to communicate with asterisk from php, there
are great examples at
http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+PHP
You'll want to use the "Originate" Action.
Moj
Andre Courchesne - Consultant wrote:
Hi all,
Here is the situation. Linu
Here is one that was made for [EMAIL PROTECTED] but you should be able to use
it.
http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html
On 3/31/06, Andre Courchesne - Consultant <[EMAIL PROTECTED]> wrote:
> Hi all,
>
>Here is the situation. Linux workstation access a web
Really all that? In my experience, which is admittedly only on the blue
iaxys and not the black ones, was just provision it, unplug power,
replug power. The black one I have now I haven't had to re-provision
yet so this could be so on them
Michael Wallette wrote:
If you want to change
Hi All,
I am working with the NBS channel driver in Asterisk. If I remember correctly
there is a way to tell the NBS how to prioritize streams currently running NBS.
Is there a way to set a priority level using the current NBS channel driver?
For instance, NBS is currently playing a music
You can use a manager event, search on Google for click to dial
applications. The nice thing about the manager is that it DOES NOT
REQUIRE that you on the same machine. As long as your access list in the
manager.conf are setup correctly you can have the php, perl, ruby, ETC.
script on the webserver
I've been seeing this for a while. No clue how to fix. The source I have
from my last update says extra_log=0, so it "shouldn't" be showing this
message at all...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JR
Richardson
Sent: Tuesday, March 28, 2006
I believe that Avaya is rebranding this device for use with there new
system ( actually they bought NIMCAT) the phones are adhoc networked -
no server- ( anyway this may be why it is a hard device to purchase.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf O
Hi all,
Here is the situation. Linux workstation access a web page on a web
server (not the one running Asterisk). From that web page, we need to
initiate a dial-out on the Asterisk server and once the call is
connected, it must ring on the agent's hard phone.
Any pointers about how to i
Hi all,
Thank you for the reply.
I didn't want to bother the list too much. However, after reading I
discover I don’t have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this
transcoding done.
My set up
From [cisco (g72
I have been evaluating the Iaxy and Asterisk for the company I currently
work for, and am rather impressed with them both. Once configured, the
Iaxy is a solid device--it's pretty much an appliance at that point
(plug it in, turn it on, and leave it alone).
My only gripe is the initial configurati
Hey,
I would like in the course of dial plan logic, to trigger a separate
outbound call. If that outbound call is answered, and if that certain
key response is detected then it will bridge the incoming call to the
newly dialed outbound call.
What I want to accomplish is that when a caller d
seems that if you get that log you didn't use jitetr buffer at all.
In my opinion the latest jitter 1.2-branch is not working, the last working
seems 1.2.1 patched.
Hope Zoa could lead us to fix it.
Regards
Rosario
- Original Message -
From: "Adam Moffett" <[EMAIL PROTECTED]>
To:
Craig,
Please correct the date on your machine. Your emails stick to the top
of the list because they have a date of 6/30/2006.
Thank you,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
___
--Bandwidth and Coloc
Just got a call from a company in Warren, MI
. They recently had an Asterisk system put in by a vendor, and are having
issues which need analysis and correction. They have a tremendous sense of
urgency. They have about (40) users, and need DID’s assigned to
extensions and are having some
Hi,
I'm confused with agents and queues in Asterisk. If I use
AddQueueMember() then "show queues" shows the agents that I have
logged into the queue... however the agent ID has to be the extension
the agent is sitting at ... kinda useless for stats tracking.
If I use AgentCallbackLogin() then "sh
Hi Domenico -
> I have a IAX2 trunk between two sites (connected with an high bandwidth
> link) but sometime/often I get:
> chan_iax2.c: Auto-congesting call due to slow response
> and call is dropped (and routed on a PSTN link).
> In iax.conf, I have:
> [iax-out]
> username=iax-in
> type=peer
We're assuming you will use a T1 (or E1) for your PSTN interface. If
you're using POTS lines then there will be no information about which
number was called--you'll need a separate POTS line(s) for each tenant.
We have multiple tenants on our hosted PBX without problem.
I was given the ta
Not a lot to go on sam. What do you want to do? If you just want to play
or have very minimal requirements then get a soekris NET4801 board, CF and
install Astlinux.
http://www.soekris.com/net4801.htm
> -Original Message-
> From: sam [mailto:[EMAIL PROTECTED]
> Sent: Friday, March 31,
Kevin Smith wrote:
Hi everyone,
I have been having some problems lately with our PRI and Asterisk, or
maybe it is just me. It happens once maybe twice a day, but when some
of our customers are calling in, the phone just drops on them. I
pulled the information below from the log from one
Mar 2
Post your 'cat /proc/interrupts' for us.
Kevin Smith wrote:
Hi everyone,
I have been having some problems lately with our PRI and Asterisk, or
maybe it is just me. It happens once maybe twice a day, but when some of
our customers are calling in, the phone just drops on them. I pulled the
inf
I agree, this does work well. My 'fax' extension is right off of the
docs:
-
[faxin]
exten => fax,1,UserEvent(Incoming Fax...)
exten => fax,n,Dial(IAX2/ttyIAX)
exten => fax,n,Dial(IAX2/ttyIAX2)
exten => fax,n,Dial(IAX2/ttyIAX3)
exten => fax,n,Dial(IAX2/ttyIAX4)
exten => fax,n,Busy
exten =>
It's been a while, but I didn't think those two terms were necessarily
exclusive. Checkpoint firewalls can provide NAT, can they not?
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mimmus
Sent: Friday, March 31, 2006 7:35 AM
To: 'Asterisk
Very true, but I want to 'sell' him on the idea, not drive him screaming
to Cisco...
Not my cup of tea, I'm afraid.
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid
Bender
Sent: Friday, March 31, 2006 6:43 AM
To: Asterisk Users Mailin
Assuming your definition of 'auto attendant' is the same as my own, then
you betcha!
If I were building such a beast, I would use a different context for
each tenant that wanted a customized IVR and a public/generic one for
everyone else. You could use DID to route the incoming calls to the
prop
I have four tenorAX boxes and there were way too many options that I would
never use. Quintum has great support so use them. Only really tricky parts on
the AX box was the dialplan section needed to be blank except min and max and
the unit ships with g729 enabled which I changed to ulaw.
Dif
Hi everyone,
I have been having some problems lately with our PRI and Asterisk, or
maybe it is just me. It happens once maybe twice a day, but when some of
our customers are calling in, the phone just drops on them. I pulled the
information below from the log from one that happened. I notice w
Dovid Bender wrote:
Can Asterisk serve as an access server/gateway to
the internet?
I have the same question. If I had a PRI coming in to
asterisk can I have users dial in and have asterisk
work as a gateway to the internet ?
Dovid
Very interesting question.
if this feature is absent,
Olivier Krief wrote:
2006/3/29, John Novack <[EMAIL PROTECTED]>:
The
reality is, of course, that telephone systems have provided this
function for many years. A DSS/BLF is available on MANY so called legacy
systems, so until this function is readily available , customers that
require a rec
> Doug this is in no way an offense to you but I think
> we need to start the asterisk booze fund. This will be
> for all of us that have ups and downs in working on
> getting asterisk set up. I for one have my friend
> "Johny Walker" right by my side when ever it gets to
> me.
I'll second that.
can anyone recommend a asterisk "turn key" company.
we will need the hardware as well as tech. support 24/7.
we'll want all the goodies, voice mail, auto attendant.
we have 6 incoming pot lines (all the same number), and 40 normal
telephones.
we have no interest in changing to ip phones or the p
NICE!
On 3/30/06, Joe Dennick <[EMAIL PROTECTED]> wrote:
> I see (and like) the demo, but where can we get it?
>
> Doug Lytle wrote:
>
> > Nicolás Gudiño wrote:
> >
> >> Something like this perhaps?
> >>
> >> http://www.asternic.org/stats/demo
> >>
> >>
> >> O
> >
> > VERY
On Fri, 31 Mar 2006, Boris Bakchiev wrote:
> That's not entirely correct :)
>
> > Fax and voice on the same DID is not possible when using a second
> > application like hylafax. Because how should the two applications
> decide
> > which one accepts the call?
>
> With the help of iaxmodem (which w
Hi!
I'm here again with echo canceller problem... :-(
I think I've done everything to enable echo canceller feature, but it
still doesn't work...
Can anybody tell me if there is some error or something missing in this
configuration please?
I'm using Eicon Diva Server 4Bri.
http://www.eicon.com
Hi Avi -
> I know this is off-topic for Asterisk, but I don't know where else to
> ask: I've setup a central directory.xml file for my Polycom IP501 phones
> with a list of all the internal extensions. None of them have 1
> as I don't want to enable any speed dials, just have a list in each phone.
Dinesh Nair <[EMAIL PROTECTED]> writes:
> On 03/31/06 19:49 Wolfgang Zweimueller said the following:
>> My conclusion with Q.SIG: do not use it at this implementation
>> level. YMMV.
>
> i'll beg to differ. we've used Q.SIG successfully with an Ericsson
> MD110 for a customer in thailand.
Well,
Kai,
Thank you for the reply.
I didn't want to bother the list too much. However, after reading I discover
I dont have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this transcoding
done.
My set up
>From [cisco (g729)] > [asterisk (sip cha
Hi,
Sometimes my Asterisk displays the following error
message...
Mar 31 12:53:04 WARNING[17170]: app_queue.c:519
statechange_queue: Failed to create update thread!
Has anybody seen it before?
Thank you
Dov
___
--Bandwidth and Colocation provid
So, if you are absolutely sure that you've specified the correct T1synchronization parameters in your /etc/zaptel.conf and you still have
fax reliability issues, look elsewhere in your implementation for theroot cause.So, would you conclude that it's possible for a given T1/E1 to have incorrect T1/
Kevin P. Fleming wrote:
WipeOut wrote:
To save bandwidth I would like to stay away from using the G.711
codecs.. Does the IAXY support GSM or iLBC? I couldn't find anything in
the docs..
No. The IAXy only supports G.711 ulaw/alaw and ADPCM.
I don't know what 'docs' you were looking in, but
Hi Sebastian -
> sorry for the long debug output below. I configured Asterisk with AMP to send
> the whole number including the extensions of the callers to the called party.
> Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but
> doesn't seem to work.
>
> 033811234451 i
Looking at the TE100P I don't see it listed Q.SIG as supported. We have it
working great as PRI. Am I wrong about the Q.SIG support?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Sent: Friday, March 31, 2006 9:04 AM
To: Asterisk Users Mailing L
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