Hello,
I have a carrier that is offering me service in H.323, but their platform is
Clarent and I am not being able to connect my Asterisk box to it using the
parameters they give me (H.323 ID, GK IP and GK ID) with oh323 registering
to their gatekeeper. I have successfuly done it with Cisco GK
Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] Asterisk and clarent
Use SIP instead of H323 on Clarent and it will work
fine
--- Jorge Alayon [EMAIL PROTECTED] wrote:
Hello,
I have a carrier that is offering me service in
H.323, but their platform is
Clarent and I am
In my country payphone solutions for Call Shops are implemented using FXS
SIP or H.323 gateways that implement the Polarity reversal feature that
reverse polarity as soon as the other party answers. I have done this in
several VoIP platforms but Asterisk.
Regular Payphones and Call Shop metering
PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 11:39 AM
Subject: Re: [Asterisk-Users] Asterisk and clarent
Use SIP instead of H323 on Clarent and it will work
fine
--- Jorge Alayon [EMAIL PROTECTED] wrote:
Hello
Hello,
I'm trying to use Asterisk as a SIP PBX with H.323 trunk connectivity.
Everything works except that calls that comes from the H.323 side do not get
audio both ways.
Since the other way round works fine (calls to H.323 side), I suspect the
problem to be in the way VAD or Silence suppresion
Hello,
I am new to this list and to asterisk and going through the archive file I
did not find an answer to my problem.
I have a VoIP network working fine with multiple gateways registered to a
Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in
that network and also
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jorge Alayon
Sent: Friday, November 19, 2004 4:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper
Hello,
I am
-Commercial Discussion
Asunto: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper
Hi Jorge,
The oh323 channel and h323 channel by NuFone are different.
As far as your problem, this looks like a codec problem i had. Try to look
that way.
K.
- Original Message -
From: Jorge Alayon [EMAIL PROTECTED
?
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jorge Alayon
Sent: Friday, November 19, 2004 4:33
Hello,
I am trying a setup that is the following:
SIP Phone (Zultys) -- Asterisk --- H.323 GK (Cisco) PSTN
Any calls from H.323 GW through GK goes to PSTN, no problem.
SIP Phone registers to Asterisk, and calling to Voice Mail, No Problem.
SIP Phone to PSTN, rings normally, on the PSTN,
Another reason that people hang to it is that on some countries fax is a
legal document while e-mail is not.
Same reason why Telex is still used were e-mail is available but fax is
not (some fishing vessels to my knowledge), communication media that has
a legal status.
Legislation changes slower
) since it does not try
to register until DOMAIN has something in i,
Regards,
Jorge Alayon
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Hello,
I am implementing a small call center with 1 to 4 agents.
For some reason I don't understand, if an agent is busy, and it is his/her turn
in the queue round, asterisk gives an all destinations are busy message and
hangs up the call. Agents are SIP lines registered with an audiocodes
= Spawn extension (macro-exten-vm, novm, 5) exited
non-zero on 'Local/[EMAIL PROTECTED],2'
in macro 'exten-vm' == Spawn extension (from-internal, 8521, 1) exited
non-zero on 'Local/[EMAIL PROTECTED],2'
-- Executing Macro("Local/[EMAIL PROTECTED],2"
Hi,
Does anybody knows how to konfigure oh323.conf to allow calls comming from a
peering gateway (i.e.: Cisco 5300) which is not connected to a gatekeeper,
and also from the gatekeeper to which Asterisk is registered ?
Something like:
GK(Carrier1)Registered to:-AS5300(carrier
Hello,
Has anyone experienced a segmentation fault in asterisk for using G729
against an AS5300 in SIP ?
I'm having this problem. Also, any other codec I use gives me incompatible
media (can be read in SIP DEBUG messages).
AS5300 configured for multiple codecs, so is Asterisk.
Tried G711u/A
Hello all,
After compiling successfully Asterisk and AMPortal, I cannot make the fax
module work.
Asterisk does not start (unless I remove the modules or mark them as Noload
in modules.conf) with the following error:
Jun 3 20:55:25 VERBOSE[3328]: [app_rxfax.so]Jun 3 20:55:25 WARNING[3328]:
Hello all,
Has anybody cofigured in SIP the Audiocodes MP108 FXS in a way that each
port is an extension of the Asterisk Box ?
So each port can have it's own mailbox, etc ?
Regards,
Jorge A.
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Hello all,
Has anyone tried 8xE1 in one box using Asterisk and Digium boards ?
What is the CPU needed for sustained performance in this capacity ?
Is this affected if G.729 codec is used ?
Regards,
Jorge A.
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success with interconnecting E1 R2 argentina? I´m
having trouble with a Meridian... I can only make calls from asterisk, but
the other way arround...
Tks
Franco
- Original Message -
From: Jorge Alayon [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
I am trying to let someone know that is being called from a specified location.
For that, the command:
exten = _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm))
should let the called person hear Anounce.gsm as soon as he/she answers.
(Only calls with prefix 107 are given this
] nombre de Eric
Wieling aka ManxPower
Enviado el: Jueves, 30 de Junio de 2005 05:22 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] Dial Option A(file.gsm)
Jorge Alayon wrote:
Hello,
I am trying to let someone know that is being called from a specified
I did it rerouting Forward Busy and Forward Noanswer from Meridian to a number
in Asterisk that was a prefix+extension, and taking that as DIDs in asterisk
directly to the voicemail of the extension. Of course there was no flasshing
light on Meridian phones, but voicemail arrives via e-mail or
Hello,
I got 3 Polycom 300 phones, and upgraded to the latest firmware provided by the
reseller.
This is my first experience with Polycom and I cannot make them register in my
Asterisk Box.
I followed every advice I found (including separating [user] and [peer] in
sip.conf.
Using ethereal,
Music on hold audio should be resampled to 8 bits 16 Khz mono and preencoded,
so audio distortion is minimized in Asgerisk encoding. This theory has worked
form me on other commercial platforms, but not yet on Asterisk, because MP3s
cannot be resampled that way. If anyone figures it out,
Capacity is planned using Erlang Formulae which is a medium complexity
statistical model mainly used for voice communications trunk occupation
and switching capacity.
Some idea of bandwith usage might be obtained using the simple
calculators at www.voipcalculator.com
Regards,
Jorge A.
Erick
Has anybody tried running Asterisk on Solaris on a SUN SparcStation ?
Or maybe the alternative of running Asterisk on a Linux Distro on a SUN
SparcStation?
I am asked to do this but I think it's almost impossible work to make it
happen.
Regards,
Jorge A.
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