Displaying 20 results from an estimated 2000 matches similar to: "VOIP Provider- Callcentric"
2007 Aug 13
1
FXO Modules and Sip Outbound
Asterisk Users,
I have never done a dial plan for this scenario before. Is it possible to
have Sip Phones make outbound calls through the PSTN? What would the call
routing/dial plan would look like?
-John
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2007 Sep 06
3
Skype + Asterisk
Has anybody ever integrated Skype with Asterisk? If you have, which
software would you recommend to accomplish such a task? ChanSkype? And how
reliable are the calls? Did the DTMF tones work? Thanks in advance.
_________________________________________________________________
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treat today!
2007 Aug 09
2
Asterisk Help
Asterisk Users,
I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service.
I have two Netgear switches on my T1 router, one for VOIP and another for
data.
I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for
all data. This morning I saw this message a few times on the Asterisk
command line. The lagged cause garbled phone calls.
Is my network to
2007 Sep 20
2
xorg-x11
Greetings,
Are there any xorg-x11-devel or xorg-server-devel rpm for centos 5?
Thanks,
Barton
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2007 Oct 10
1
Re: scp -t . - possible idea for additional parameter
>> I understand that that is not how scp works today.>And it will likely never change.
Why not? Just because "That's how we've always not done it" doesn't sound like a very good reason to me.
>> I'm suggesting that we make a minor change to how it works.>scp is maintained for compatibility reasons only, as I've understood>things.
That's still
2014 Apr 14
1
how to configure callcentric peer
On 11.9, trying to set up a callcentric peer:
sip debug:
> <--- SIP read from UDP:204.11.192.161:5060 --->
> INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0
> v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d
> f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127
> t:
2008 Jul 28
2
Callcentric Issues
Hey,
I have a few dids with callcentric. They seem to work fine most of the
time but at some points I get "handle_request_invite: Failed
to authenticate user <sip:PSTNnumber"
This happens intermittently.
The way I understand it the insecure=port,invite should tell asterisk
not to authenticate users coming from that host. But its not working for
some reason.
This is my sip.conf
2014 May 23
1
Way off topic: gvoice and callcentric
To deal with google dropping xmpp for voice, I've gotten a callcentric
number. The cc number connects to asterisk, and all works fine. Then I
set up the cc number as the gvoice forwarding number. If I'm on the
gvoice site, I can make a call and it will ring my cc number and then
the outside number. That also works fine.
BUT, when an outside call comes into gvoice it forwards the call
2007 Oct 02
3
scp -t . - possible idea for additional parameter
How difficult would it be to add an additional parameter to the -t that would *lock* the user at that directory level. say -T instead of -t...
By locking, I mean translating /path/to/file as ./path/to/file, or ../../../path/../../../path/to/file as ./path/to/file.
Basically set a root point as the current home directory, then build the pathing based on that, any "../" would become
2007 Sep 13
5
CallWithUs Service?
Asterisk Users,
I am thinking about selecting CALLWITHUS as my sip provider. Has anybody
ever used them? How was the call quality? DTMF Tones issues?
Thanks in advance.
-John
_________________________________________________________________
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2007 Oct 31
2
Sluggish throughput with htb
All,
I have been using the following as a means of rate limiting access to the Internet via eth0 (which connects to my cable modem) and it was working great with my 2.4.20 kernel:
tc qdisc del dev eth0 root
tc qdisc add dev eth0 root handle 1: htb default 1
tc class add dev eth0 parent 1: classid 1:1 htb rate 486kbit ceil 486kbit
tc qdisc add dev eth0 parent 1:1 handle 10: sfq perturb 10
2008 Nov 18
1
setting up callback
Greetings Asterisk users!
I'm trying to setup Asterisk system to act as a callback system together
with callcentric (http://callcentric.com) but it appears that I hit common
DTMF issue and I want to workaround this problem. Basically my current
setup is the following:
1) I have dedicated Asterisk server that it is linked to my callcentric
account
2) I have US phone number (DID) from
2007 Aug 08
3
VoicePulse Connect
Asterisk Users,
Has anybody use Voicepulse Connect for Asterisk?
I am trying to cover all my bases because in the past, I got burned with
poor quality of service, along with failed DTMF tones with 3 different SIP
Providers (Vitelity, Broadvoice, and Teliax).
I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP
protocol. Any insights would be great. Thanks.
-John
2019 Feb 28
3
Asterisk - can't hear other side. Or other side does not hear us
Antony,
It is correct. Noone connects to Asterisk box/server from outside.Callcentric SIP trunk configured and Asterisk maintains connection to it itself. No special ports opened, nothing. Connection happens from us to Callcentric and all calls routed in from CallcentricI don't know exactly how it's doing it by it works.
Again, keep in mind it is working for many years for most / 90+% of
2009 Jan 17
1
Sip Trunk registration
Hi
Can anybody help me on this ?
I am using Asterisknow 1.5.0-Beta(Freepbx)
I am having a problem getting the sip trunks to register.
It makes no different which provider one is using.
Trunk name: callcentric
Peer Details:
context=from-pstn
fromdomain=callcentric.com
fromuser=1777xxxxxxx
host=callcentric.com
insecure=very
secret=pasword
type=peer
username=1777xxxxxxx
Register String:
2008 Apr 01
4
Voicemail- Recorded Mesage Low Volume
Asterisk Users,
I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian "Etch" system. On the recorded voice mail messages, the volume is really low when retrieving them with my cell phone. I tried with multiple cell phones with the volume level high and still, the same problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it had no affect on
2018 Oct 16
2
Is there any way to pass caller id to
Thanks all,
I did contact Callcentric about it and their tech support helped meget those headers established. They even helped to troubleshoot Asterisk dialplan.
A the end all works as it should.
Thank you,Ivan
Message: 2
Date: Mon, 15 Oct 2018 23:39:31 +0200
From: Daniel Tryba <daniel at tryba.nl>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at
2018 Nov 16
2
Queue not dialing out to cell phone for some reason
My settings for the queue.log are in the [general] section of logger.conf
I'm running 13, I didn't see what version you said you were running.
If I wanted to add a LOCAL channel to my queue I'd do it as
member => LOCAL/7124 at kiniston-intern,0,John,hint:7124 at kiniston-intern
On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch <idemkovitch at yahoo.com>
wrote:
> John,
2007 Aug 02
6
Teliax Quality of Service
Asterisk Users,
I recently ran into some problems with the quality of service with Teliax.
This occurred on August 1, 2007 with a dropped outbound call, audio
quality isse on the callee side- not hearing me well on callee side, and
sending DTMF tones (configured for RFC2833). Am I the only Teliax customer
having this problem?
It seems like when I am ready to go live with my Asterisk
2008 Oct 13
0
Support for CAF in flac command-line?
RF64 support sure would be nice, but it wouldn't really help to do
this "instead of" CAF. For one thing, Logic Studio Pro does not seem
to support RF64, because the manual states that WAVE and BWF are
limited to 4 GB. CAF may be a format which lacks universal support,
but RF64 is also very limited in usefulness. Treating either one as
a substitute for the other is not