help me to resolve this issue.
Regards
Vikas
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org
and Kerberos/PKINIT. Does it support all the
encryption mechanisms as recommended by PC-1.5 Specification.
3. Can it act as an announcement server.
Regards
Vikas
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Hi Asterisk Developers/users,
I am facing a problem while using the cdr feature of asterisk(version
asterisk1.4.24.1). Whenever I make a call using a “*.call” file and it gets
failed , it don't produce the CDR for that channel as it falls into
“OutgoingSpoolFailed” channel As there is no such
in the 916 area code for bandwidth.com
But other providers like www.broadvoice.com are able to provide us
VOIP services in the 916 area code.
I am wondering how can I get the bandwidth.com service,
Thanks,
Vikas
___
-- Bandwidth and Colocation Provided
I need 100 DID's in a specific rate center (916-854-). How do I go
about finding who owns the rate center ? If the DID's are available in
this rate center ?
Thanks
Vikas
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
you should consider spoofing the number (IE calling using XXX-XXX- and
presenting as 916-854-).
But if I spoof the DID the person receiving the call will not be able
to get back to me. So I do not think that is going to work for me.
Vikas
On Wed, Feb 25, 2009 at 12:56 PM, Danny
Spoofing the caller id is not an option for me. I am wondering how do
I go about buying the DID's
Thanks,
Vikas
On Wed, Feb 25, 2009 at 2:16 PM, Danny Nicholas da...@debsinc.com wrote:
So they are going to (eventually) make a legitimate (in some cases) practice
Illegal because of spammers
caller id. So that the person receiving the call can call back
directly to the caller id that they received on their phone instead of
going through the IVR hell.
Vikas
On Wed, Feb 25, 2009 at 3:13 PM, M Hulber asterisk-ad...@hulber.com wrote:
Since it's not clear from this thread of conversation, do
. When people call in on the 800 number take the local number they
are calling from and then call them back from our unlimited outgoing
account from broadvoice.
2. Find a vendor with a better rate.
Any idea what we can do to better manage the 800 cost.
Thanks for your time,
Vikas
server and then to the
asterisk server in CA ?
Since if that was the case it would add a 60ms lag time to the conversation.
Thanks for helping me clear this up,
Vikas
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
, what equipment is
it on? It's not the switch, as the switch is Layer 2.
On Fri, 13 Feb 2009 10:09:03 +0100, Benny Amorsen
benny+use...@amorsen.dk
wrote:
Vikas topg...@gmail.com writes:
The ISP said that they ran a fiber optic wire to a media box at our
office and from there there is a RJ45
?
Thanks once again for such valuable feedback.
On Fri, Feb 13, 2009 at 8:21 AM, D Tucny d...@tucny.com wrote:
2009/2/13 Vikas topg...@gmail.com
My questions are:
1. The black wire coming into the Mc Manstel box is that a fibre optic
cable ?
2. What is the Mc Manstel box doing ?
3. What CISCO
The ISP giving net access at our office has installed a 24 port CISCO
2950 switch in our server room. I can buy 24 connections from them and
get 12Mbps of Upload but each individual connection is restricted to
512Kbps.
Currently we have requirement of 20 simulataneous calls so we
purchased 4
...@gmail.com wrote:
On Thu, Feb 12, 2009 at 4:04 PM, Vikas topg...@gmail.com wrote:
The ISP giving net access at our office has installed a 24 port CISCO
2950 switch in our server room. I can buy 24 connections from them and
get 12Mbps of Upload but each individual connection is restricted to
512Kbps
12:41:51 Jeff LaCoursiere wrote:
Get a Cisco with five ethernet ports. Use one for your connection to
asterisk. Use the other four as your connection to the ISP, and MUX them.
Great way to spend 5K :)
j
On Thu, 12 Feb 2009, Vikas wrote:
I have asked the ISP to rate limit a single port
Observation: When the agents talk using the polycom 650 handset the
voice quality on the other end is much better compared to if they talk
using the plantronics noise cancelling headset. If you would like a
recorded phone call when a agent is taking using the headset vs when a
agent is talking
I need SIP trunking from a high quality business service provider for
25,000 SIP minutes growing at approximately 10% each month.
Currently we are using exgn.net to provide inbound 800 (Costs $200 for
approx 10,000 minutes a month)
and we are using broadvoice.com for outgoing calls (Costs $100
Concise summary: When I set the TOS to Minimize-Delay the DSCP field
in the packet changes from Expedited Forwarding to Unknown
Here are the details:
Scenario 1: IpTables is not used to set the TOS
This is what the packet looks like using wireshark:
Internet Protocol, Src: 59.93.192.xx
Requirement: Monitor the QOS for the SIP phones connecting to the voip server.
Ideal solution: Browder based reporting software that I can install on
the asterisk server (I use freepbx) and when I connect to this
reporting engine it gives me the Jitter loss, packet loss and latency
for each of
center in the basement
and has DS3 coming into the data center. I can buy as much bandwidth
as I want from the data center.
Regards,
--
Vikas
http://www.stanford.edu/~vikask/
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
. complete product line with conference phone and receptionist phone
(not on Aastra)
3. cordless (not on 501/430)
4. backlit LCD (not on 501/430)
5. Inbuilt POE (not on 501)
6. speaker phone
7. 2 network ports.
Which one will you choose ?
Vikas
If I commit to Aastra what do I for the:
A. Receptionist
B. Conference room
Aastra does not seem to have a phone for these two functions.
Any suggestions ?
Vikas
On 1/22/07, Jay Moore [EMAIL PROTECTED] wrote:
IMO, the 480i, by a LONG shot.
The 480i is easier to use, looks nicer, has better
How to make one touch record on asterisk 1.2.1 use mixmonitor app ?
In res_features.c line line 469:
monitor_app = pbx_findapp(Monitor)
How to make pbx_findapp return mixmonitor ?
T
___
--Bandwidth and Colocation provided by Easynews.com --
I am authenticating sipura device as a sip user to my asterisk server.
Things work fine and then suddenly asterisk console tells me:
Oct 26 23:09:17 WARNING[5096]: chan_sip.c:4826 check_auth: Stale nonce
received from 'Sipura1PSTN sip:[EMAIL PROTECTED]'
as soon as that happens if i try to call
user=x
username=x
Why is this happening,
vikas
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
Does any one know of starvox? and when would you choose starvox over teliax ?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
I need the quote please. Would appreciate a off list quote.
thanks
On 8/28/05, Chris Felter [EMAIL PROTECTED] wrote:
Please send me a quote for remote installation of
Asterisk, GUI administration, and billing for calling
card, caller ID based prepaid, and postpaid.
Off list please.
something?
Thanks
Vikas Deolaliker
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I read a few discussions on installing Zaptel modules in Fedora Core 2 with
2.6.5 kernel. I was wondering if there is a definitive FAQ on this? I am
still unable to install by FXO card in my pbx box because the modules won't
install.
Thanks
Vikas
29 matches
Mail list logo
