Displaying 20 results from an estimated 60000 matches similar to: "ipkall & one way audio"
2010 Sep 16
4
one way audio for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
2004 Nov 22
1
Using IPKall and SIP with insecure=very
Hi all,
I've got one of those cool free incoming IPKall phone numbers from
www.ipkall.com. These numbers just connect to the SIP proxy of your
choice, they default to Frreworld Dialup. You can use them with your own
sip proxy on asterisk. My config for this is below.
The trouble I'm having is the incoming calls do not seem to hit the
section in sip.conf for the call. With sip
2009 Sep 19
0
IPKall using iax
Is it possible to receive a call via IPKall through IAX connectivity without registration?
If so how to set it up.
I've run-into and old link;
http://forum.voxilla.com/ipkall-support-forum/ipkall-beta-testing-iax-connectivity-without-registration-26728.html
--
Joseph
2004 Feb 03
2
IPKall->FWD->Asterisk
Hi Folks,
I recently setup an asterisk system in order to provide a telephone phone system for my web hosting business at a very low expense. My problem is that DTMF tones are not being recognized when calling the IPKall phone number. Calling my server via FWD and IAXTel works out fine however. Has anybody experienced this with the IPKall service? are they not passing the DTMF tones or am I doing
2004 Oct 03
3
ATA's
Hi, Has anyone had any luck using modems on ata's other than with Cisco
ATA-188's? I really don't have the money pay for the 188's as this is for
my personal use.
Thanks.
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2006 Apr 30
1
integrated voip originator, to digitize audio once and only once?
Calling 7777 from a local extension on my local
network, I get good voice quality from asterisk, and
asterisk reliably recognizes my dtmf input.
I set up a sipphone trunk (free) and called in to it
via a separate sipphone account on another computer,
and got slightly lower, but still good, audio quality.
I set up a FWD trunk (free) and called in from the
other computer, and got somewhat lower
2009 Apr 06
2
IPkall
Does IPKALL still exist?
I am after a free SIP trunk - who is still giving these away these days?
As I noticed Stanaphone is no longer in business?
Regards,
Dean Collins
Cognation Inc
dean at cognation.net
<mailto:dean at cognation.net> +1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London in-dial).
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2005 Feb 01
1
Re: Asterisk-Users Digest, Vol 6, Issue 325
> Message: 1
> Date: Fri, 21 Jan 2005 17:38:27 -0600
> From: "Henry Devito" <hdevito@qwest.net>
> Subject: [Asterisk-Users] SPA-2000
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@lists.digium.com>
> Message-ID: <000d01c50012$4ea49f30$4300000a@homeacxa7jw2xn>
> Content-Type: text/plain; format=flowed;
2004 Sep 30
1
Queue Setup almost got it
Check my reply to your last post.
Use SetGroup and Checkgroup before sending the call to your agents.
Robert Jackson
-----Original Message-----
From: Henry Devito [mailto:hdevito@qwest.net]
Sent: Thursday, September 30, 2004 10:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Queue Setup almost got it
Ok I think I have the queue
2009 Sep 15
0
1.6.2.0-rc1 intermittent voicemail problem ?
1.6.2.0-rc1. I am having trouble with voice mail intermittently not
working correctly on CHANUNAVAIL. (it may happen for other statuses
too, haven't checked). Basically here's what happens:
-- Executing [1651xxxxxx at mydids:1]
Macro("SIP/ipkall-trunk-14838bc8", "phone,1651xxxxxx") in new stack
-- Executing [s at macro-phone:1]
2004 Dec 08
7
more then two wildcards in one machine
Has anyone had successfully installed more then two digium wildcards in
the same machine?
I'm going for four.
thanks
Shoval Tomer,
IT Manager,
SofTov Advanced Systems, Ltd.
Office: +972-3-9230686 ext. 179
Fax: +972-3-9216642
Mobile: +972-54-8000200
2009 Aug 20
12
IPKall and FWD
We all know the FWD is NO more available.
How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite
?
Any alternative for FWD ?
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2004 Sep 27
0
Re: Asterisk-Users Digest, Vol 2, Issue 281
Now that most of you have worked overtime to show why most people are
continually pissed at Nix Users (all except two of course). The problem I
can see is the downright technosnobbery involved. There is nothing wrong
with Linux. I play around with RH9 and FreeBSD and find that most things
run fine. But you get into a problem where it keeps asking for the same
blamed libraries over and over on
2010 Jul 22
0
SIP URI Dial has one way audio
Hi,
I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ
as target user which is registered.
Asterisk server IP: 70.118.x.x
calling user IP: 117.58.x.x
called user IP: 117.58.x.x:5062
First I dialed my registered user in normal way like this,
Dial(SIP/XYZ,30,rtT)
and during conversation audio was OK in both ways. Then I dialed the
registered user via
2005 Jan 05
0
One way audio [Asterisk + Innovaphone IP3000 + asterisk-oh323/h323]
Hello everybody,
I?ve been trying to solve a problem for several weeks now but it really
beats me.
There are several hard phones connected to an Innovaphone 3000 VoIP gateway.
On the other side I have a SIP softphone connected to Asterisk. The problem
I have is that on incoming calls (hardphones to softphone) I only have
outgoing audio (from soft to hardphone); everything is OK when I call the
2004 Sep 30
2
Queue Setup
Hi, I am on my next venture now, Need to set up 3 queues. I would like
these setup using the agentcallbacklogin. Does anyone have an example of
this? I have looked through the wiki , but all that did was confuse me.
One of the problems I'm having is how do I configure my extensions.conf to
dial the agentcallbacklogin
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2010 Dec 24
1
One way crappy audio in iax call - Asterisk 1.6.2.15
Hi,
We had 2 asterisk 1.4 connected together in iax, all was fine. One of
them was upgraded (server and Asterisk) in 1.6.2.15, the other end is in
1.4.38
When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But
calling from 1.6.2 to 1.4 give a bad audio to calling party (words are
cutted, you can't understand the words). On callee party it's still good.
We replace
2006 Nov 06
1
Audio goes one way during the call for a fewseconds. Is it RTP, NAT, dyndns, or what it is?
We had very similar problems to this which drove us insane for ages.
Basically we use VoIP trunks (SIP) for all our inbound + outbound calls.
Call quality was good however we would get random problems where people
could not hear us or us hear them for about 5-10 seconds at a time.
After weeks of trying to get to the bottom of the problem it appeared
our VoIP trunk provider (sentiro/sip2go) had
2005 Jan 21
5
SPA-2000
Hi, I have not implemented any of the spa-2000's yet. Do they work ok with
asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is
it two fxs ports with the same extension?
2006 Nov 03
4
Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?
Hi everybody,
I finally want to get rid of 1-way audio problem. Please help me here.
I have 3 scenarios.
1. Audio is always one way. Caller who dialed can't listen the called party
but called party can listen him. In this scenatio Asterisk is on dynamic IP
with dyndns FQDN. sip.conf has externip = abc.dyndns.org and localnet =
xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. Where is