hi,
i would prefer a Mediant 1000 with 12 ports FXS of Audiocodes to do the Job.
further info available upon request,
Mickey
On 7/12/06, Mike [EMAIL PROTECTED] wrote:
Hi,
I`m looking for a SIP-PSTN adapter, basically to switch a customer from a cheap PBX to mine, but resuing their own
hi,
you can try the following:
exten = s,1,Gotoif($[${CALLERIDNUM} = 1130851536 ]?10)
exten = s,2,Goto(from-pstn,s,1)exten = s,10,disa(no-password,from-internal)
it works for me.
if you need further help, let me know.
BTW, i am very interested in the Brazilian Market so i would like to get as
hi Ronald,
i would use a CallerIDNum authentication, based on the Asterisk Database to solve it.
then you do not need any verification.
you just build a list of approved numbers in the database and then have a context checking the whitelist.
if you need more help, let me know,
Mickey
On 4/8/06,
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:
hi Ralf,
AFAIK, the 486 CANNOT do what you want it to do.
you need a call thru functionality and this can be achieved only with 488 .
then you conect the line to the old analog PBX extension and any call coming to this extension will enter the Asterisk as a call from a SIP extension.
i hope i
Dmitry,
it seems to me that just in the definition of the extension, (Outbond CID - thru the AMP) you just define the CID of that extension.
be carefull to give the proper CID, within your block, to that extension.
good luck,
Mickey
On 3/29/06, Dmitry Ivanov [EMAIL PROTECTED] wrote:
On Tuesday
as of now, 8 ports for 8 phones.
there will be soon a 16 ports version (within April)
On 3/25/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
How many phones lines ?
-Message d'origine-De:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] De la part de Curt ShafferEnvoyé: vendredi 24 mars
i would suggest Astribank-8 of XorCom. it is a dedicated Asterisk compliant solution.
you can look at : www.xorcom.com.
On 3/24/06, Curt Shaffer [EMAIL PROTECTED] wrote:
Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations?
hi,
if interested please consider the TigerNetcom box of 104 for doing the same functionality, much better piece and at considerable lower price.
for technical information on how to use it i would be happy to assist off list.
On 3/20/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Mar 19, 2006, at
Aaron,
of course it is possible. r u using AMP? if positive, look atcustom extension.
there you can define a forward application so using the proper process , you can have there the required forwarded destination
all the best,
Mickey
On 3/10/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Does anyone
hi Zach,
i would use GOTOIF to forward the DID from within the [incoming] context to the other context. i would try :
exten = gotoif($[did]=DID1,goto did1|s|1,)
exten = gotoif($[did]=DID2,goto did2|s|1,)
On 3/4/06, Zach A [EMAIL PROTECTED] wrote:
Both DIDs are SIP and from the same provider.
Conrad,
i would go with following solution:
1.6 sets of Audio Codes of 24 FXS ports conected by SIP accounts to the system. the type is MP 124. then you open the conector on the initial MDF and then the users have the same phone on their table
2. one dual Xeon system (or even stronger - 2 Dual
hi Vikram,
this is quite simple if the incoming call has DID.
if it does, then
1. in the AMP extension define the IP:port as a SIP extension.
2. using the AMP you go to the INBOUND Routing. enter the DID you want to route to the IP:port.
3. enter the extension numer you have defined in step 1
Arne,
my idea is as following:
in principle, i would do it in different contexts so in everycontext we have a different extensions.
i would be happy to get some example if such principle can be implemented.
Mickey
On 3/1/06, Arne Morten Johansen [EMAIL PROTECTED] wrote:
Hi there.Is it possible to
i think we can help. we do have there some contacts and if the total volume would be significant, we can give a nice quotation.
let me know off list what you exactly need.
BTW, $0.23/minute is much much high compared to our solution.
On 2/28/06, Johnathan Corgan [EMAIL PROTECTED] wrote:
Sam
hi all,
maybe i am mistaken but it seems to me that the HiPath 2000 series is an Asterisk based system.
why am i saying this? because Siemens announce it is a Linux, Open Source system.
so, as i do not know any OTHER PBX Linux- Open Source system rather then Asterisk, does anybody know something
hi ,
i have some options we are working with at vast deployement with no problems:
www.tigernetcom.com - type 102 is a nice ATA, like GS 486 but far away better.
we import directly from the producer at great prices so if anybody interested, please contact off-list.
additionaly, we know another
Manoj,
just look in AMP to Inbound Routing, fill in the DID, define the softphone as extension X and send the call to extension X
Mickey
On 2/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,I am a newbie to Asterisk and I was able to install Asterisk and call out.Recently I purchased
hi Palma,
as the SJ initiate the call, it will allways go with GSM Codec as the codec should be identical used on both sides. as you do not have G729 on the SJ, it will never use G729.
furthermore, i think that if the GSM will not work, then the second option choosed would be PCMA
i hope i gave
Hi Sam Tam,
i would be interested in these ATA that you can offer.
please provide me with more details about this option.
thank you very much,
Mickey Lazar
On 2/9/06, Sam Tam [EMAIL PROTECTED] wrote:
We have got some ATA for only $55 if you are interested?Sam-Original Message-From:
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