try using Frig.. it's a great client with an SIP client.. i tried it on IPhone
and on my N82 Nokia phone.. it works great on GPRS and Wi-Fi... and i DO use it
with my two Asterisk Servers..
regards
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
for smart phones?is it frig or
fring?
On Oct 3, 2008, at 11:49 AM, Tariq .. wrote:
try using Frig.. it's a great client with an SIP client.. i tried it on IPhone
and on my N82 Nokia phone.. it works great on GPRS and Wi-Fi... and i DO use it
with my two Asterisk Servers.. regards
AHD Tarek Sawah
: Re: sip
clients for smart phones? Tariq, fix your email client (it eats line
breaks in text/plain). And please leave out the parts which are not relevant
(list policy). Thanks. Philipp Kempgen --
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma
GmbH - Bachstr
: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial Plan
Issues On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote: this
is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read
the dial plan!! What is the dialplan? ls -ld
] Dial Plan Issues
Steve Murphy wrote:
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote:
Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of
my servers is not allowing the other
: [asterisk-users] Dial Plan Issues
This is a better question asked on a Fonality list. Maybe they have a
manual.Thanks,Steve Totaro
On Thu, Sep 25, 2008 at 10:21 AM, Tariq .. [EMAIL PROTECTED] wrote:
Greetings,i have two asterisk servers running on Centos with asterisk 1.4.21
and trixbox..i
per day. Agents will be available but calls will
just sit there until one of the waiting agents logs off and back in.
Andrew
Tariq .. wrote:
the Autofill thing didn't solve the problem.. i have another server
hosted in the USA with Asterisk 1.4.20-1 on it.. it doesn't have
Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of
my servers is not allowing the other to send calls while it is possible in the
opposit direction..
i have the same exact
The NAT Network Address Translation is a layer three protocol... it
encapsulates the End user's IP address with the router's IP address... so your
Asterisk is not recognizing the IP address of the end user.. if you are
insisting on using the HOST option with a specific IP.. maybe you should
between 2 asterisk Hi Tariq, Tnx for your reply. Tried adding the
deny/permit but still gave me the same result. I still have these error,
handle_response_invite: Failed to authenticate on INVITE regards,
nhadie Tariq .. wrote: im not sure this will help but i did the same
settings
im not sure this will help but i did the same settings you mentioned and added
my lines and it worked..
you need some sort of authentication between the Asterisk boxes.. and the
easiest way to do it is to do it like this
[asterisk-2]type=peerhost 10.20.30.2 *** i will assume that you have
I have LinkSYS PAP2t and it worked the way you discribed it.. Asterisk simply
assigns a different port for the peer automaticaly.
Date: Wed, 20 Aug 2008 20:09:32 +0100 From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Subject: [asterisk-users] Two peers, same IP
and port Is it
Hello
I have a SendTEK UNIK-22 USB ISDN TA unit attached to my Asterisk
i it possible to use it to make and receive calls with asterisk? and if so can
anyone help me? or at least give me some hints? i tried but couldn't manage it
_
Greetings..
My client is starting a new business and they required a strange thing.. they
want an IVR system that can be integrated with some competetion.. the scenario
is he folowing
the caller VoIP Provided will reach the system.. the IVR picks up and
welcomes them .. multiple language
my best offer to you is to read more about the dial plan to understand what
happens.. or try to understand what does freepbx do and what does it write and
understand the applications..
Date: Wed, 23 Jul 2008 20:53:45 -0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Hello are you using FreePBX for your configurations? there is an option in the
extentions.conf for queues called
CWIGNORE=TRUE
try disabling it and see if it works for you .. this is the best i can help you
with .. i am using call-limit combined with busy-limit to stop the call
waiting.. i
Try adding busy-limit=1 to your SIP users as it will let the agent to report
the Busy as a hint.
the call-limit=1 only allows one channel to the agent.. but then if the agent
is not busy the queue will try to call them and it will bypass the CW service
so the Agent channel will receive the call
the only suggestion i would have for you is to use a SINGLE file for your MOH
.. and you record the welcoming note in the begining of the file.. so whenever
a caller comes in .. they will hear the MOH .. which has the welcoming note
before the music starts...
i know it's a stupid trick but it
i have been through this with Dynamic agents and callback .. i used to use
addQueueMember but it caused me troubles when joining a queue.. because
sometimes the agent would be in the queue twice..
my suggestion to you is to check if you can make calls between two members of
the queue .. then
Greetings..
i have 20 extensions with two queues.. i have members in the queues as SIP/
now recently i have noticed that users are unable to call each other.. this is
causing me a headache..
calls comming to the queues are forwarded smoothly to the users.. but they
can't call eachother..
Greetings..
i'm having problems when a queue uses RINGALL stratigy ..
so i am thinking of FOLLOWME to do the work ..
now my question which is best to use?? i have 5 groups of 10 users each..
I use Asterisk 1.4
regards
Tarek
_
i think what you have is a Codec problem.. i have faced similar problem with a
remote phone when internet was not able to carry G711 which is used by defaul
so chage your codecs to G723 or GSM and it will work fine..
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Fri, 20 Jun 2008 15:17:01
Greetings..
i'm having a disconnection problems with Calls comming to my Call Center..
i'm using the free version of G729 .. and i'm starting to suspect it would be
the reason.. i just need to know if it's possible or there will be other
problems??
i asked the technician in the location to
Hello
My Flash Operator Panel keeps resetting timers everytime i open it or refresh
it..
is there a way or config to force it to maintain timers ?
_
It’s easy to add contacts from Facebook and other social sites through Windows
Greetings.. i'm facing a slight problem i hope.. the management in my call
center requires using the chanspy 555 to monitor newly hired agents.. and there
seems a problem where the monitoring extension gets stuck and can't soft hang
upit .. anyone got a solution for that? it just gets stuck
you can reduce the 5 seconds to any number you wish.. but from a personal
experience .. if you put the retry to zero.. nothing will change.. i suggest to
use 1 as your minimum aiting number
Tarek Sawah
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Wed, 4
Jun 2008
?
Best Regards,
Ahmed N Tariq
RADIOFRAME NETWORKS
www.radioframenetworks.com
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