Displaying 20 results from an estimated 2000 matches similar to: "to china: good voip service providers?"
2005 Jun 27
1
Re: teliax [Was: LiveVoip is Bankrupt]
For outbound only, I have traditionally recommended VoipJet. They just
recently has a spat of issues that seem to have resolved though. I am
still using them via their east coast server and it seems to work quite
well again. Cost is around 1.3 cents minute I believe. Use IAX and
g711 for best quality to VoipJet.
Thanks,
Wiley
-----Original Message-----
From:
2005 Jul 28
3
Cisco Call manager
Anybody using Cisco Call Manager and connecting to any SIP termination
service like voipjet, voxee, etc? Please msg me offlist.
AK
2011 Mar 01
2
two questions regarding incoming call
Hello,
I want to make an agi script to match incoming DIDs with usernames.
I tried to do such entry in incoming trunk.
[DID_diddw]
include = from-didww
[from-didww]
exten = 3130XXXXXXX,1,AGI("did.php")
exten = 3130XXXXXXX,n,DIAL(SIP/${yup_no},20)
but when i run the rule it says
chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to extension
2006 Nov 09
5
Voxee lag problems ?
Anyone having problems with voxee since last few days or is it just me ? In
peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency
. Most of time it is 20 ms or so but when i start sending traffic to them
latency increases to 1000 ms or even LAGGED ( also shows high in peak time
even when no high latency ). No problems with any other provider . Anyone
else having same problem
2005 Oct 02
0
Console Sound: Cuts out, Comes back after restart
I'm having a problem with sound output to the console.
My basic dial plan is as follows:
exten =>
_1NXXNXXXXXX,1,Dial(IAX2/####@voxee/${EXTEN},30,A(beep))
exten => _1NXXNXXXXXX,2,Playtones(info)
exten => _1NXXNXXXXXX,3,Hangup
I get the following output in the console:
___*CLI> dial 1#######@voxee
-- Executing Dial("ALSA/default",
2005 Oct 03
0
Console sound output -- shuts off when call from console answered
I've got a problem with audio output from the Asterisk console. I'd really appreciate any help.
I'm simply trying to dial out to a phone on PSTN. My extensions.conf entry is as follows:
exten => _1NXXNXXXXXX,1,Dial(IAX2/####@voxee/${EXTEN})
exten => _1NXXNXXXXXX,2,Hangup
After starting asterisk and dialing, I hear a ringback tone through the console speaker, and the PSTN
2005 Sep 11
0
extensions.conf for VOXEE using SIP!!
Hello,
I have been trying to setup a Voxee Sip termination. If anyone has
extensions.conf different than Voxee suggestion.
Can you please send me a copy?
Thanks!
Jerry
Voxee web site advises to use:
[voxee]
exten => _1NXXNXXXXXX,1,Dial,SIP/${EXTEN}voxee
exten => _1NXXNXXXXXX,2,Hangup
exten => _011.,1,Dial,SIP/${EXTEN}voxee
exten => _011.,2,Hangup
2006 Mar 19
0
Bizzare DTMF on channel bank
I have incoming PSTN lines on an Adtran 750 channel bank. Calls are
evaluated by an agi script based on callerid and forwarded to an
international DID through Voxee. There is an IVR at that number that
asked to user to enter a selection. When the user presses a key, my pbx
puts the call on hold and tries to start music on hold. What's doing
this? I have no backgrounds, no listen, the call
2006 May 09
0
Using ChanIsAvail and SIP
I am trouble finding a configuration that works for ChanIsAvail and SIP.
My two providers are Voxee and Teliax.
I have these lines in a macro
exten => s,n,ChanIsAvail(SIP/teliax&SIP/voxee)
exten => s,n,Cut(CH=AVAILCHAN,-,1)
exten => s,n,NoOp(AVAILCHAN= ${CH})
; Dial the available Channel
exten => s,n,Dial(${CH}/${ARG1},60,t)
Looking at the execution, I can see what the AVAILCHAN
2006 Jan 22
1
Fail over using CHANAVAIL
I am trying to construct a macro for long distance dialling. I have two
internet feeds, I have all routes including Teliax on Internet A and a
static route to Voxee on Internet B. I thought I could use the dialplan
entry below which uses the ChanIsAvail() command to check the
connection, but this returns the provider but not the username, so I
don't understand how to use this for real
2013 Jun 15
0
Freight forwarder & logistics provider shared an album with you.
Dear My Friend
Nice day, Hyun Young is a leading professional freight forwarder
and logistics provider who focus on the shipment from South China to all
the world. Hyun Young started freight forwarding operation at Shenzhen in
2004. Based at Shenzhen, our ambition have pushed us forward to expand to
other cities in south of China. Now we have capacity of handing shipment to
or from all
2006 Jan 28
0
AutoDialing with VOP USING SIPURA 2100'S
Hello all,
I am trying to find out if anyone has a provider that is good with dtmf
playback using a Sipura 2100? I've just dialed with voxee and the call goes
through but when I press 1 my dialer does not " hear" it.
My dialer is making the call using a Dialogic d/4PCI connected to the
Sipura 2100 through voxee and I am calling my landline. When I pick up the
landline
2006 Jan 27
2
VOXEE Caller ID..
I cannot find any means of passing my own Callerid using Voxee. It always
comes across as NO ID, or nothing, or unknown.
I could not find anything on their website about setting your own caller
id in the system either. (their web account pages).
Is anyone here using their own Callerid information through Voxee?
thanks
2006 Apr 21
2
confused about iax and voip providers termination
Hey guys,
I'm actively trying to get the "big" picture on how all this works and
relates to each other.
I've gone through some basic examples from the book and from the sample
files just fine.
Now, I've setup an account with a VOIP provider which does IAX termination
(exgn.net)
After getting an account and following their steps, I can make calls out
using my IAX (cubix) and
2013 Jan 03
0
recycled ink cartridge for office stationery
Dear Manager,
This is greetings from Amy (ShenZhen 4inx Technology Ltd.,) an ink cartridge
manufacturer with more than 8 years experiences,I'm very confident that our
quality can meet or even exceed your expection.
Now we are working with many clients in Europe I hope we could have
opportunity to work with you in the near future and I believe you will be
impressed by our
2005 Jun 09
2
Sixtel is still alive?
Whoa, talk about flying under the radar. I got a few DIDs *months* ago
from Sixtel (or iax.cc). Initial respone was great, but then it seemed
that the only tech-support person had fled the country. No responses,
bad responses, poor call-quality. I had a few $s left in the balance
and kind-a just forgot about them. Not worth 10 minutes of my time to
get $10 of my money back, quite frankly.
2006 May 23
1
Configure Voipjet.com content in Asterisk
Hi,
I am Chandramouli from India. I have successfully implemented Intercom (Dialling within my office LAN) and Voicemail features using Asterisk. To implement this, I am using X-Lite Softphone.
Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any external hardware to implement pure VoIP solution.
Here I am sending my configuration file values:
Contents of
2005 Feb 16
0
More jitter buffer questions
I've been trying to resolve some quality issues and I was hoping
someone might be able to provide some insight.
To give you an idea the calls are coming in via a SIP DID and sent out
via an IAX2 connection. Latency to both the SIP equipment and IAX
equipment are around 80ms with 0 packet loss accoridng to ping tests.
The server is located in a data centre so bandwidth is not an issue.
Most
2008 Sep 28
1
Dream of a wiki GUI for R
Dear R fans ( and wiki fans),
I am just writing a draft to introduce confidence intervals of various
"effect sizes" to my students. Surely, I'll recommend the package
MBESS in R. Currently, it means I have to recommend R's interface at
first. As a statistics teacher in a dept of psychology, I often have
to reply why not to teach SPSS. Psychologists and their students hate
to
2005 Jan 15
2
IAX2 Channels & Bandwidth
Hi all,
I'm using VOIPJET to make international calls with an IAX2 connection
between my local asterisk server and their server(s).
At times I seem to have a problem if 5 or more international calls are made
at once - I'm on a 1024kbps download and 256kbps upload DSL line (only the
asterisk server uses this DSL line). Today I switched the codec from ulaw
to ilbc in an attempt to lower