Thank you for the reply, actually we are looking for something like the
followinghttp://www.ebay.com/itm/GSM1SIP-GSM-over-IP-GoIP-SIP-Quad-Bands-voip-gateway-Quad-band-1XGSM-GoIP-VoIP-/181075100268how
ever our requirement are a bit wire like SMS in addition to Call capability.
Tarek Sawah
Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel (one
SIM card). any suggestions?
Tarek Sawah
--
_
-- Bandwidth and Colocation Provided by http://ww
20:38:22 +
> Subject: Re: [asterisk-users] SET SIP_CODEC and Video issues
>
> Of course you are disabling the video maybe also include the video protocols
> in the sip_codec
> -Original Message-
> From: Tarek Sawah
> Sender: asterisk-users-boun...@lists.digium.com
aw)
exten => _.,2,Set(SIP_CODEC_OUTBOUND=gsm)
;exten => _.,2,Set(SIP_CODEC_INBOUND=gsm)
exten => _.,n,DIAL(SIP/TK${EXTEN})
exten => h,1,Hangup()
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RH
send him calls. He has been advised to use Sippy which they
claim is more stable than Asterisk.
i'm not an expert with Sippy so i'm looking for a piece of an advise here.. if
i'm doing an Asterisk Vs Sippy comparison. can anyone help?
Regards
Tarek Sawah
Information Tech
i like asterisk -rx 'show channels concise'
give less detailed but more readable output
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> From: da...@debsinc.com
> To: asterisk-users@lists.digium.com
Hello,
Is L6 a remote device? is there any firewall residing between the server and UA?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> From: panyc...@gmail.com
> Date: Tue, 25 Oct 2011 14:30:53 +0300
> To:
are on the same internet link it doesnt' happen to all of
them at once.. only one of them.
i suggest trying to change ISP for testing.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
From: ak...@abacus-it.
One more thing can you post your peer's configs as you have it in the config
file? and can you register with the same user from within the lan?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> Date: Sun
=
then we will be able to check your problem?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> Date: Sat, 15 Oct 2011 19:08:10 +0200
> From: ad...@tootai.net
> To: asterisk-users@lists.digium.com
> Subject: R
what version of Asterisk are you using?
try issuing "agi show" from the Asterisk CLI console and see if you get some
output?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> Date: Wed, 12 Oct 2011 1
i had a similar challenge having Asterisk listen to multiple ports.. "some of
my agents located in countries where SIP is blocked"
the only effective way is to use IPTABLES i believe your problem can be solved
with the same method.
Tarek Sawah
Information Technology Adviser
if i'm not mistaken you want a way to know who is busy and who is not in some
way of a live monitoring method?
why not try the Flash operator panel? ir have a look at the code of it's server
and see what you will be needing to use
Tarek Sawah
Information Technology Adviser
can you post the while dialplan? it seems cropped somewhere as i dont' see it
starting or ending anywhere.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 5 Oct 2011 12:31:49 +0500
From: govoi...@gmai
i think you can try placing the beef file in the /var/lib/asterisk/sounds
directory and not the language specific one.
and your system is calling the beep file without having it in the dialplan?
sounds strange somehow to me.
Tarek Sawah
Information Technology Adviser
Integrated Digital
Google is your best friend when looking for this type of assistance my friend.
try callcentric vonage packet8 for reliable retail DIDs.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Sat, 1 Oct 2011 00:51:59
those features won't be of use for
him for a card that will allow him to talk as much minutes as he can! you
abusing "free routes" or not.. is not his business actually.
those features can be offered to PINLESS customers who can pay 100-300 $ per
account!
Tarek Sawah
Information
, allow DID sales.
those issues have more effect on your business.
could have helped in US DIDs.. but in Asia i'm no aware of the presence of such
providers. however TATACOMMUNICATIONS is the largest VoIP Operating entity in
that region and you may find some luck contacting them?
Tarek
What does (international long) mean exactly? are you a calling cards company?
if so you should look for some company that will be charging you like 0.004
Cents per minute.. and you can find companies that will add more channels to
your DID.
Tarek Sawah
Information Technology Adviser
month per did.. 5 Euros per month
and you should pay Extra for Extra channels.. could be the same amount for the
same amount of channels
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> Date: Thu, 29 Sep 2011 11:09:
one adjustment i would suggest is using (|) instead of (,)
exten => 222,n,Dial(SIP/${EXTEN}||KkTtL(6))
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 18:32:28 +
From: salah.elha
an
over do and playing such sounds files at this rate will consume the resources!)
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 18:22:57 +
From: salah.elharit...@gmail.com
To: asterisk-us
have a look at the following:
"L(x[:y][:z]): Limit the call to 'x'
ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is
required, 'y' and 'z' are optional."
source
http://www.voip-info.org/wiki/view/Aster
for you.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> From: alexreca...@gmail.com
> Date: Wed, 28 Sep 2011 18:59:39 +0200
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Receiving mu
this is related to your carrier's SIP messages as they are sending a sendonly
attribute instead of sendrecv (taking a wild guess here) your asterisk will act
as if the call was placed on hold thus the MOH butts in.
an sip debug log for a similar call will be more helpful?
Tarek
g the
call and calling back again"?
what is the timeout for the agent setup in the queue settings? or more helpful
if you paste your queue settings
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
the internet to send packets back.
this is one of the scenarios i can think of. and can be done in 20 minutes.
well it can be expensive if you calculate the costs of an additional computer
on the network. :S
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RH
(G729).. and it works
with WIFI.. i use it at home.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> Date: Sat, 27 Aug 2011 10:14:24 +0100
> From: gordon+aster...@drog
i did do some Asterisk tests on SUN VBOX .. works like a charm but you need to
dedicate some good resources to the virtual box!
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> F
you didn't provide your "dialplan" for the incoming call context from_poland?
nor registration string?
could be a dial plan problem .. or codec issue.. as long as you register
"properly" the server has no problem with NAT.. it's a routing or codec issue i
th
whole db or a part or it in order to do it's manipulation.. so i'm not sure if
the database will be "locked" by one of the asterisk boxes when writing to it?
which prevents the rest from writing to it at the same time?
regards
Tarek Sawah
Information Technology Adviser
Actually i had to upgrade to 1.6 due to a provider problem with session-timers
and RTP data .. then i downgraded again to 1.4.
do you suggest that i test 1.8 instead of 1.6?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492
calls.
my question is .. is there a different in resource consumption between all
versions? how come 1.4 could handle over 500 calls while 1.6 crashed at 100?
please advise?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
at : http://elastix.org/
regards
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> Date: Mon, 12 Sep 2011 05:14:08 -0700
> From: bilmar...@yahoo.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [as
if you provide what kind of reporting you need it would be easier to point a
few pointers?
either you can build it yourself.. or try the Call Center module from Elastix..
can be a good tool
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
because they are behind a router and using private IP addresses. and the Cisco
router is Nating our traffic
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> From: sat
this is happening on all Soft phones are facing the same problem. Zoiper ,
X=lite , our own pjsip based dialer (CRM).
this was not the issue .. it happened suddenly .. we switched internet links
even.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE
an be provided.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
--
_
-- Bandwidth and Colocation Provided by
Greetings
i've setup a new asterisk server 1.4.38 ... everything works fine however i
need to register the server with another SIP provider..
the registration string ..
the server is not attempting to register .. sip show registry shows nothing..
i created an sip_registration.conf file and as
etail not wholesale, you may
understand my question more clearly now?
Regards
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> From: sherwood.mcgo...@gmail.com
> Date:
If you look at it the way you want it.. you usually tell your customer the
available funds and minutes in their account right?
How will you explain "politely" that you have dropped their calls for lack
of balance because someone else used their account?
If you don't tell them their balance and call
i think it's SIP_CODEC now .. and not _SIP_CODEC?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> From: da...@debsinc.com
> To: dan...@tryba.nl; ast
Has any of you tested Vyatta Load balancing and fail over solution with
Asterisk? It uses heartbeat and works like magic with regular traffic but
didn't have the time nor chance to test it with VoIP traffic.. but I think
it's the same way.
Anyone?
-Original Message-
From: asterisk-users-bo
DID you grant your user the ability to INSERT into the MSSQL db?
I have asterisk inserting easily
Just a privileges issue
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand
Sent: Sunday, Septem
rces.
Just give it a try and let me know.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> From: basit.e...@gmail.com
> Date: Sun, 26 Sep 2010 02:43:05 +0500
> To: asterisk-use
our billing
system.. C# application interacting with Asterisk doing all the math. after all
it's all SQL and Asterisk working. you can do that with a dial plan i believe..
so why not build an AGI to do it for you?
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoI
A quick answer? A2billing.
It has what you call it differential billing.. but they call it progressive
billing.. 3 steps .. for 3 different rates ..
Go for it.. easy to setup and quick to learn and use.
Regards
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@li
A quick answer? A2billing.
It has what you call it differential billing.. but they call it progressive
billing.. 3 steps .. for 3 different rates ..
Go for it.. easy to setup and quick to learn and use.
Regards
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@li
i don't see any mistakes in your question.. but i still don't get it.
what do you need exactly from Fax on demand? sending faxes? receiving faxes?
From: zoelha...@yahoo.co.id
To: asterisk-users@lists.digium.com
Date: Fri, 24 Sep 2010 17:27:57 +0700
Subject: [asterisk-users] Fax On Demand - Aste
Bilal,
If you are using 3G or Wifi with your Nokia Native SIP Client.. try to
connect via an internet connection sharing machine.. it seems that your ISP
is blocking INBOUND SIP packets.
Test and let me know
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-
tion.. can we add two of those cards to the server? Will it be
efficient?
Regards
Tarek Sawah
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar e
Gareth
Usualy the queue has the ability to know if the agent is "INUSE" and skip
them.. you can simply use ringinuse=no to the queues.conf under the queue
itself or the general section and that's it .. no need for the whole
dialplan.. as you are using SIP members.
Salam
-Original Message-
TOMER
ALREADY IN DATABASEexten => _1N.,n,Hangup
you can use the above example to check the number being dialed against your DB
(what ever DBMS you are using) and route it depending on the result of your SQL
query.hope this helps
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoI
can you state your internet connection your agents are on?and one more thing..
how are the members positioned into the Queue? static? Dynamic? single station
and call forwarding (find me follow me extension in the queue)? do you get call
waiting override with Auto Answer?
-- Tarek Sawah
) this can be
caused by a configuration of the queue itself something related to memberdelay
directive. try setting it to "0" or something similar.Regards
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 386 492 9993
From: william.stillwell-li...@ableb
ty is not
guaranteed .. but i need to restrict the agents to their seats and my CRM
software
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
> From: asterisk_l...@earthshod.co.uk
> To: asterisk-users@lists.digium.com
> Date: Tue, 29 Jun 2010
Lets say you did everything as it was mentioned in the tutorial .. then go into
Asterisk console and issue the command:sip show peer A2BILLINGCREATEDUSER
if you can't find it.. then simply include additional_a2billing_sip.conf in
your sip.conf file.Regards
-- Tarek Sawah
Integrated Di
nd get the sip username and password assigned to
him and use it through Zoiper or any other softphone to make calls ..our agents
are allowed international calls .. so we want to restrict them to only use our
dialer.Is that possible?Asterisk version 1.4.33regards
-- Tarek Sawah
Integrated Digit
m?I'm asking as i'm looking
for a similar setup just trying to set it up virtually before we go live.Regards
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
> Date: Fri, 25 Jun 2010 11:49:12 -0400
> From: j...@ngn-networks.com
or US companies providing DID
numbers to US citizens without FCC license.
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
Date: Wed, 23 Jun 2010 23:43:14 +0530
From: rscl.mum...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Nee
their provider have a limit of 30 minutes per call ..
so the caller had to redial.. unless it's automated.still you can provide us
with more info.Regards
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 386 492 9993
> Date: Wed, 23 Jun 2010 16:08:51 +0
IP sections XXX no one will
assist you as no one will know who is talking to whom.. just like if you go to
a doctor with a prostate problem.. you can't tell him that you won't remove
your clothes off ;)regards
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 38
didforsale.com is one of the best and reliable DID providers in the USA
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
Date: Wed, 23 Jun 2010 16:50:48 +0530
From: rscl.mum...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users
d Database .. you can build your own reports and so.
or you can use a2billing to do the billing and ACD.. Elastix has a good billing
(without a2billing) .. but i prefer a clean installation of asterisk and work
around with database and PHP much better.. Good Luck!
-- Tarek Sawah
Integrated Digit
what do you mean unblock the calls exactly?
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
Date: Fri, 18 Jun 2010 11:12:55 +0100
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk issue
Hello
does that phon has a static IP? does it register with the server? posting your
SIP.con and extensions.conf related to this issue could help us to understand
what you are doing.
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
From: niksingha
when you add an agent to a queue the agent should log in try adding
member=SIP/301member=SIP/302instead of agent directives.this will ring both
phones.. from your output it doesn't seem to be ringing the agents at all.
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP US
users and
passwords then assign a2billing accounts to them to make it safer.. plus the
fail2ban .. give it a try.
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
> Date: Sun, 13 Jun 2010 22:28:38 -0700
> To: asterisk-users@lists.digium.com
>
it's autofill=yes i'm sorry for the typo
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
> Date: Mon, 31 May 2010 11:33:09 +0200
> From: mass...@archivio.it
> To: asterisk-users@lists.d
a portion of your quues.conf and you sip.conf pasted can be helpful? try using
autofull=yes in your queues.conf and see if it works
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
> Date: Mon, 31 May 2
Greetings List,Trying to interconnect with a new provider.. the require
a compliance with RFC 3261 so knowing less than needed about RFC
documentations.. i would like to know if Asterisk is actually in compliance
with RFC 3261 or not.. Can any one help with this?
Regards
--
Tarek Sawah
the simple way i can see it is the following;let's say you have did starts
with 1708
[from-did]exten = _1708XXX,1,Answerexten
= _1708XXX,n,Queue(SALES,,)exten = h,1,Hangup
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 34
SDP is
> probably sent with
>
>
>
> a=sendrecv
>
>
>
> I believe your server is acting correctly.
>
>
>
> -Original Message-
>
> From: asterisk-users-boun...@lists.digium.com on behalf of Tarek Sawah
>
> Sent: Fri 4/30/2010 12:11 PM
>
> To: Ast
;tag=as00522e07
To: ;tag=gK02b3c8db
Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z
CSeq: 102 INVITE
Contact:
Allow:
INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 0
<->
[K --- (9 headers 0 lines) ---
[
goes well with no problem). and let me
take a wild guess.. your provider is offering a premium number services.my
advise check your internet connection on the remote location and keep a ping
from that network to your server running all the time to check for time outs.
-- Tarek Sawah
Integrat
at is going on?
--
Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562
2308
_
Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox.
Simply place the SIP Extension of the GSM gateway in another context
context=from-gsm
and in your extensions.conf use something like this
[from-gsm]
exten= => _X.,1,Goto(whatever IVR you want)
> Date: Mon, 26 Apr 2010 17:23:40 -0300
> From: aco1...
gt; cover their a$$ as the route was down and it wasn't blocked.
>
>
> I doubt the problem is with sending calls to different media gateway as I
> think SIP signals take care of that. Just like canreinvite feature. But I
> reserve the right to be wrong.
>
> -Bruce
>
&
you got the name EXACTLY!
i already am doing what you suggest but facing problems with some destinations
and they claim that the problem is with my Asterisk server not their routes!
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347
Greetings list
i'm trying to connect with a VoIP provider for termination.. and they have
offered us three servers to connect with
one SIP Signaling server and Two Media servers ..
googled for a week and didn't find a way to do this.. so my question. is it
possible to be done?
Asterisk server
s.
my experiences are with small call centers up to 40 seats ..
Thank you for your help and support.
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562
This is the first time i face this issue..
i have an extension 100 .. calling 0018001234567
is there a way in Asterisk to get info that 100 is calling that number?
sorry for the lame question but i never had to know such info on my system.
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA
gents
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
From: jonas.kell...@telenet.be
To: asterisk-users@lists.digium.com
Date: Wed, 18 Nov 2009 16:21:12 +0100
Subject: [asterisk-users] Queues without agent login
for the sake of bandwidth you are supposed to connect each two servers
together.. otherwise calls between B && C will have to go through A .
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
From: i...@saudihom
o
hit the HTTP API
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
Date: Mon, 9 Nov 2009 22:19:08 -0500
From: thomas.per...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SendText
Will text messag
you need to post you SIP.conf and your Extensions.conf so someone can have a
look at them and see if there is anything missing
what are the contexts you are using with your peers?
what is the dial plan triggered when calling your destination number?
--
AHD Tarek Sawah
Integrated Digital Systems
the registration string will solve the issue for
me.. and i think it will do the same for you.
regards
--
AHD Tarek Sawah
> Date: Mon, 10 Aug 2009 12:55:41 +0200
> From: patr...@erdbeere.net
> To: asterisk-users@lists.digium.com
> Subject:
been testing with Sun VirtualBox and i managed more than 30 extensions on a
2GHz Dual core machine with 1 GB ram for the VBOX.. just not running recodring
or encoding .. things went well
--
AHD Tarek Sawah
> Date: Fri, 7 Aug 2009 08:47:03 -0700
>
times you will have to dial again for
the call to get setup.
regards
--
AHD Tarek Sawah
> Date: Thu, 6 Aug 2009 22:59:40 -0700
> From: spamsucks2...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Anyone had
ly send calls with codec G.729
my agents get best voice quality with G.711u
I need your advice .. am i missing anything in this setup?? it used to work ..
and it STILL works on another hosted server with Agents located in Morocco..
with a different vers
rs strictly send calls with codec G.729
my agents get best voice quality with G.711u
I need your advice .. am i missing anything in this setup?? it used to work ..
and it STILL works on another hosted server with Agents located in Morocco..
with a different vers
on is not supposed to be directed to this list then just disregard
it my friend
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
> Date: Tue, 4 Aug 2009 07:32:15 -0400
> From: abalas...@evaristesys.com
> To:
terisk behind vyatta and configured the nat .. system up and
running smoothly ..
if anyone else have tried it please let me know if any problems have been faced
Regards
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
do you suggest buying a licensed Software from Digium?
Date: Sun, 2 Aug 2009 18:53:16 -0400
From: stot...@asteriskhelpdesk.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk and E1 Cards
On Sun, Aug 2, 2009 at 6:37 PM, Tarek Sawah wrote:
Greetings List
Greetings List,
i have a new question regarding Asterisk and E1 Cards
a client of mine is requiring an Asterisk Server with 2 E1s.
the scenario is the following
they want 400 extensions to register with the system.. and required 64
concurrent calls.
added to it that they are expecting the system
some how the extension you have identified in your extensions.conf file is
wrong..
you are forwarding your call to an extension @ a local extension??
you can try at least the following
[default]
exten => _X.,1,Dial(SIP/${ext...@proxy.sp.co.kr)
it may work .
let me know
--
AHD Ta
i'm not so familiar with what youa re talking about .. but i beleive i've seen
something like that in FreePBX where you can setup a failover trunk for a
context.. try to have a look at it. and i hope it's what you are looking for
--
AHD Tarek Sawah
Integrated Digital Systems
C
accountcode is a setting you add to your SIP peer.. so it doesn't require
restarting Asterisk.. only restart the SIP module..
"sip reload" will be enough my friend..
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA:
from it like any
other SIP calls inside the server..
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
> Date: Sat, 16 May 2009 14:46:27 +0300
> From: timotsm...@gmail.com
> To: asterisk-users@lists.digium
try adding callerid=CIDNAME
this will force your callerID in your DIalplan
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
> Date: Wed, 29 Apr 2009 09:38:58 +0300
> From: oguzh...@bilkent.edu.tr
> To: aster
1 - 100 of 113 matches
Mail list logo
