Displaying 20 results from an estimated 2000 matches similar to: "VoipTalk unable to accept calls at present?"
2004 Jun 16
1
VOIPTalk silver service
There was some discussion on this list recently about the voiptalk silver
service. I've just had an e-mail from them saying that the price has been
reduced to 2.99 per month. However, they still only provide an 0870 number
whereas pipecall provide a local call rate 0845 number in the fee.
Chris
2004 Dec 20
7
'I'nvalid extension handling problems, even with workaround
Hello folks,
I'm having trouble configuring Asterisk to play an "invalid extension" message to
anyone dialing an undefined extension.
First I tried using the 'i' pseudo-extension, but it didn't work at all;
searching the wiki I found that page:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension
where it basically says that the 'i'
2007 Jan 22
4
X100P how do i recieve incomming calls?
I have just purchased a 2nd hand X100P, if I do a ztcfg -vv I get:
"Zaptel Version: 1.4.0
Echo Canceller: MG2
Configuration
======================
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured."
My understanding of the above is that the zaptel driver has detected the
card. What do I now need to do, in order to get an incoming call to work
with
2004 Jan 26
1
SIP behind NAT - use of "externip" option
I am having difficulty configuring SIP behind NAT (using latest CVS).
Using sip.conf:
[general]
port=5060 ; Port to bind to
externip=ww.xx.yy.zz
bindaddr=0.0.0.0
nat=yes
register=>[userid]:[password]@voiptalk.org/2000
[voiptalk.org]
nat=yes
externip=ww.xx.yy.zz
type=friend
secret=[password]
nat=yes
reinvite=no
canreinvite=no
I fail to register. SIP Debug gives:
SIP
2004 Dec 23
1
Problems with incoming IAX calls...
Trying now to set up the final part of my * switch. I must admit I've had
great fun over the last week or so playing with it, and would like to thank
you guys for all the assistance (past and present, since I've been trawling
a lot of old posts!!!).
Scenario - using voiptalk.org to supply the incoming gateway, tied to an
0845 number for convenience in testing. Internal 7960 -> 7960
2004 May 22
1
Sip proxy registration help
Hi All,
I have just installed Asterisk and am trying to connect it to a SIP
account that I currently have with www.voiptalk.org but without any
success. Although I know that voiptalk do provide asterisk accounts I
don't want to convert the SIP account until am happy that it's gonna
work for me. The asterisk box is currently behind a firewall and the
following ports are being forwarded
2005 Jan 04
3
voiptalk.org IAX service - user experiences
Hi,
Anyone used this service, any comments on reliability/support?
Thanks
John
2004 Aug 30
1
IAX.conf problem (NEWBIE ALERT!)
I have several of incoming numbers on IAX from voiptalk and magrathea
but have a problem with IAX.conf. If I follow the example from voiptalk
[VoIPTalk Incoming Number]
type=friend
username=VoIPTalk Incoming Number
context=[XXXXXXXX]
and make incoming entries in IAX.conf for the numbers like below with a
different entry for each number pointing to a different context,
incoming numbers always
2004 Dec 22
2
Why use 'Answer'?
Why is it that newcomers always feel like inserting 'Answer' is a
necessary step in their extension.conf entries?
>[voiptalk.org]
>;forwards any calls starting with an "8" thru voiptalk.org
>exten => _8.,1,Answer
>exten => _8.,3,SetCIDNum(55555555)
>exten => _8.,4,SetCIDName(My Name And Surname)
>exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)
2004 May 18
1
Configure asterisk for outgoing.. need authuser parameter?
Hi,
I have access to two providers. On one of them the authuser is the same as
the username, so outgoing works. On the other one I can only get
incoming -
what ever combination I try for outgoing I get an error. The register
command
has the ability to specify both usernames (which is why incoming works) but
outgoing doesn't seem to, and without that I'm stuck.
They are defined as:
2005 Feb 21
1
NAT-helping outbound proxy
Hi,
We're deploying a small VoIP solution for a group of teleworkers.
Naturally, this exposes us to all sorts of fun, most of which we seem to
have working properly. However, some NAT issues are still bugging us and
we have noticed that often these situations didn't exist when users were
connected directly to our VoIP provider, voiptalk.org.
They have something which they call a
2010 Jan 09
1
UK dialing tone
Hi,
I use VoIPTalk as my provider and unsure of a minor issue. When people call me they get a US ring tone instead of UK. Is this a Asterisk configuration issue or one for VoIPTalk ? I am running 1.6.2.0 and IAX2 trunks.
Thanks - Phil
2005 Jul 11
2
DTMF not sending properly via IAX
I'm not sure if this is a -users or a -dev question, since the answer
probably comes down to something in the code.
I'm running the latest CVS-STABLE, and am subscribed to PSTN service
using IAX2 via Voiptalk in the UK.
I've just been alerted by a customer that the sending of DTMF from my
asterisk box to a remote PSTN user doesn't work, although it used to.
To test it, I have
2005 May 18
2
Call forwarding...
Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based -
2004 Jun 23
1
Codecs and pauses
Hi all
My * implementation is working brilliantly with only one small fault left to
kill.
I'm using IAXTalk from Telappliant for my incoming/outgoing calls to the
pstn network; if I set my codec to GSM everything works great - no pauses
but quality is a bit poor. If it set the codec to alaw (I think I'm using
the correct one - I'm in the UK) I get intermittent pauses on the call.
2006 Apr 17
4
Looking for a good VoIP Provider in the UK-
Any recommendations for a VoIP provider in the UK?
I have a few guys in a field office in the UK with SIP phones and a VPN
tunnel back to a working Asterisk setup in the US. The Asterisk setup
has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US
offices, so they can call vendors, customers etc in the US at local
rates. I'd like to get the same thing for the UK, so that UK
2004 May 27
5
Silly incoming SIP failure
Hello folks,
i upgraded to the actual CVS head from yesterday (27.5.) but can not get
incoming SIP calls from my provider (sipgate). If someone calls my
number, my asterisk responds with the following error:
May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request:
Failed to authenticate user "<CallerID>"
<sip:<CallerID>@217.10.66.11>;tag=as38e9693c
I
2004 Aug 01
2
Zaptel - incoming delay
I am new to Asterisk so can I start by apologising if this question has been asked and answered already.
I'm in the UK using BT for two incoming lines, one on Wildcard TDM400P and the other on Wildcard X100P. I also have a SIP connection to voiptalk.org.
Incoming calls via SIP/broadband ring on extensions immediately. However, an incoming call via PSTN is displayed on the CLI as an
2004 Apr 21
6
Help choosing a UK IAX provider
Hi,
Currently using voiptalk.org and the quality is getting really bad.
I would like a second provider preferably in UK, anyone got any
suggestions?
Ta.
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2004 Nov 18
3
SipTone II
Anybody used the above phone with asterisk
I have one working ok for calls, but having a problem with voice mail.
Using either the 'Voice mail function key' or dialing 88 (for my system)
just gets me to Call Terminated
Asterisk CLI shows the error message 'unable to get User name'
My Grandstream works ok, asking for User name, then Password
Any ideas ?
--
Clive
Email :