Kannel + AGI seems like the best solution as long as you have a SMPP
connection. Has anybody been able to send MMS thru Kannel? (Integrated with
Asterisk, of course). I know there's MBUNI for Kannel but is there a Gateway
out there for people with no SMPP connectivity?
Jesus Mogollon
On Tue, Mar
I'm also experiencing the same problem. Has anyone found a fix for this?
Jesus
On 2/7/07, Santiago Aguiar [EMAIL PROTECTED] wrote:
Hi everyone!
I'm using Asterisk 1.2.7.1 on a CentOS 4 server with 5 - 9 agents and I'm
having some issues with the Chanspy application. All the agents are on
server. I'm using
1.2.13 (heavily patched) and I came across your email. Any chance of getting
your port? Thanks for your help...
Jesus Mogollon
On 2/22/07, Craig Guy [EMAIL PROTECTED] wrote:
Hi Richard,
there was a thread regarding this a while ago on the dev list which
resulted
in a patch being
Let me rephrase that:
It would be a lot *simpler*. From the programming point of view you are
connecting to a single events source. Astmanproxy is very stable as well.
On 3/3/07, Stefan Reuter [EMAIL PROTECTED] wrote:
Jesus Mogollon wrote:
The best option would be to use AstManProxy
The best option would be to use AstManProxy and connect your event manager
to it.
Jesus
On 3/2/07, Doug Garstang [EMAIL PROTECTED] wrote:
Ok, so I ain't much of a Java programmer, but...
Can the Asterisk Java API be written with threads? Ie, I need to connect
to multiple Asterisk systems
Hi all
Does anyone know of any motherboards with PCI slots that can take the
TE412P card? Is there such a MB for Athlon 64 or P4 procs?
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]: chan_unicall.c:4216 setup_unicall: Unable to register channel '1-15'Jul 24 22:50:44 WARNING[9252]: chan_unicall.c:4536 reload: Reload of chan_unicall.so is unsuccessful!
Why would this be the case?Jesus Mogollon
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This patvh allows you to do just thathttp://bugs.digium.com/view.php?id=6681On 5/5/06, A.J. Paxson
[EMAIL PROTECTED] wrote:Hi all!Curious if there was a way to introduce multiple announcments in
the queues.Rather than just:Thank you for holding.Your call is important to us...wait 60
yum install libidn-develOn 4/19/06, Luis Fernando Ramírez Cueva [EMAIL PROTECTED] wrote:
I am instaling asterisk on Fedora core 3. I have instaled
zaptel-1.2.3, libpri-1.2.2, but when I am instaling (make install) asterisk I have the following error:_GNU_SOURCE -O6 -march=i686
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMailIt's all in thereOn 2/13/06,
Tomislav Parčina [EMAIL PROTECTED] wrote:
Hi list!How to send a call directly to voicemail recording?When I put thisexten = 313,n,VoiceMail,u221Or thisexten = 313,n,VoiceMail,b221In my dial plan,
Don't use the file extension (get rid of .pub when defining the key in dundi.conf). It will work.Jesus MogollonGlobal IP Systems, LLChttp://www.globalipsystems.com
On 2/1/06, Jonathan k. Creasy [EMAIL PROTECTED] wrote:
I am getting the following message when trying to lookup up a number
-port T1 cards in this machine (dual xeon, 1 gig of RAM).
Voice calls are perfect. I was reported that, under the old system,
everything worked perfectly (also an asterisk system, albeit a very old
one). Has anyone hit a wall like this? Any help will be *REALLY*
appreciated!
Jesus Mogollon
for this is definitely not Q.421. That spec does not mention DTMF at all. R2 using DTMF doesn't appear to be in the ITU specs, as far as I can tell. Without a spec, or any equipment to play
with, there isn't a lot I can do right now. Steve Jesus Mogollon wrote: Hi Steve:Thanks for your help. I really
in unicall.conf set the amount of channels you need in a group and then call
Dial(Unicall/g1/${EXTEN})2005/11/10, Bruno de Assumpção Loureiro [EMAIL PROTECTED]:
Hi users,how can I do a group of unicall channels to round? Is there somethinglike Dial(Unicall/R1/...) ??Sometimes a outbound channel
... 2005/11/4, Steve Underwood [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED]: Jesus Mogollon wrote: Does anyone know how to make this work with Asterisk? (R2-Digital (Q.421)) I have MFCR2 configured but I'm told that outgoing calls are
to use Q421 R2 Digital signalling. Any help is appreciated. Jesus
Hi all
When I try compiling libmfcr2 I get:
spandsp.h: present but cannot be compiled
Any ideas?
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Does anyone know how to make this work with Asterisk? (R2-Digital
(Q.421)) I have MFCR2 configured but I'm told that outgoing calls are
to use Q421 R2 Digital signalling. Any help is appreciated.
Jesus Mogollon
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me an error. Any ideas? Thanks for your help...2005/11/4, Steve Underwood [EMAIL PROTECTED]:
Jesus Mogollon wrote:Does anyone know how to make this work with Asterisk? (R2-Digital(Q.421)) I have MFCR2 configured but I'm told that outgoing calls areto use Q421 R2 Digital signalling. Any help
Well, that's why you'd see languages refered to as
ES_mx
ES_es
ES_pe
ES_ve
And so on much better way to deal with language variants...
2005/11/4, Andres Tello Abrego [EMAIL PROTECTED]:
Mexican Spanish..Ha, funny term...:)Mexican Spanish = mx from MeXico...es = ESpain...So, es would be..
Hi has anyone used MFCR2 using Unicall? I need to use the
protocol_variant=fx but Asterisk crashes saying that there isn'ty such
a module, though it appears as an option in the configuration file.
Does anyone know why it isn't working?
Jesus Mogollon
http://zarzamora.com.mx/asterisk/48
I know it´s a tutorial for E1/R2 but it has a new patch that works with CVH-HEAD and spandsp. Try it out.2005/10/31, Andy Kuo [EMAIL PROTECTED]
:Hi all,
I'm trying to install spandsp. I followed the instructions on
Hi All: I'm in need of a phone that would blink a led to let the callee know that there is an incoming call. The GXP-2000 does this but I want an alternative to Grandstream. Any help is appreciated.
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and use Polycoms or, if too
expensive, maybe Sipuras 841. These GXP-2000s are pure evil.
Jesus Mogollon
Global IP Systems, LLC
http://www.globalipsystems.com2005/9/1, Aaron W [EMAIL PROTECTED]:
I have upgraded the GXP-2000 to the newest firmware 1.0.1.12 and the
phone is much more usable However
start to stutter and ringing starts
becoming less and less loud until barely a beep). Thanks for your help.
Jesus Mogollon
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AFTER we upgraded to the new
firmware.
Has anyone seen this?
Jesus Mogollon
Global IP Systems, LLC
http://www.globalipsystems.com/
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Hi Lee:
NTP is working as expected, but it does take a couple of minutes (!) to get the date from the server
Jesus Mogollon
2005/8/25, Lee Archer [EMAIL PROTECTED]:
Hi, do you have an on-site NTP server? I found that
after the firmware update NTP from the * server stopped
working
Greetings all!
I bought some Polycom 501s and got them installed, configured
and running. I'm using bootrom 2.6.2 and sip 1.5.2. My problem is that
when I press the Do Not Disturb button, the phone stops responding and
reboots by itself eventually. Has anyone seen this problem before?
Jesus
Asterisk + Kannel. When you need to send a message, call the kannel
directive with a System call. This is assuming you can connect to a
SMSC via SMPP.
Jesus Mogollon2005/7/13, Ronald_Wiplinger [EMAIL PROTECTED]:
I am thinking of SMS and wonder if I can set-up with Asterisk a SMSC anduse SMS to
Hi...
has anyone written or seen a variation of Saydigits that behaves like
Background (listening and responding to DTMF)? If there's such a beast,
I'd sure like to know... if not, how hard would it be to implement?
Jesus
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How about Xlite for Linux? It's already out... www.xten.com2005/6/27, Hamish Whittal [EMAIL PROTECTED]:
Hi Folks,I am wanting advise on a good soft-phone on Linux. I have looked at
Gnophone but cannot seem to get it to compile under debian sarge. I amnow looing at sipXphone seem to be picking up
). How do you register an agent? The documentation is
lacking...
Jesus Mogollon
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Hi all:
Has anybody used ny of the predictive dialers for
Asterisk? I got the assignment of installing one, so I'd like to get
any opinions beforehand. I'll post my experiences once installed and
up-and-running! Thanks!
Jesus Mogollon
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Hi all
I installed asterisk on a dual PIII 700 with two NICs. I then
proceeded to configure both NICs with bonding enable (bonding
miimon=100 mode=1). I know certain features (like load balancing) under
a bonded configuration is not understood by some switches, so I
configured it using mode=1
So am I (sorry to drop in like this). I'm a programmer and I'm open to
start a project like this based on this attempt. Let me know.On Apr 8, 2005 9:46 AM, Douglas Conrad [EMAIL PROTECTED] wrote:Spencer,I am interested for your asterisk manager.Can you send for me?[]sDouglas ConradG.Marshall
Greetings.
Has anyone used dialparties.agi (part of AMP)?. In my setup, every
once in a while, the script monopolizes the CPU driving it to a 100%
usage and won't terminate. has anyone seen this problem?
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