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Home > Archive > Voice Over IP in UK > March 2006 > sipidiscount.com - any good
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| Author |
sipidiscount.com - any good
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| paul woolley 2006-03-19, 11:22 am |
| is it ?
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"paul woolley" <junk@webvista.co.uk> wrote in message
news:e3cTf.58703$zk4.15258@fe3.news.blueyonder.co.uk...
> is it ?
>
I've had no issues with outbound calls. Very good value for money and
quality.
I buggered if I can get inbound to work with Asterisk.
That's assuming you mean Sipdiscount, and not as you've put in your subject,
sipIdiscount
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| Martin² 2006-03-21, 2:45 am |
| Works pretty well, specially on VoIP / QoS router, occasionally need to
re-dial, but that no bother.
The only gripe I have that they now take your money after 120 days whether
you used it up or not !
Now you have to consider it as a quarterly rental :-(
Regards,
Martin
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| M.Dexter@blueyonder.co.uk 2006-03-21, 2:45 am |
| On Sun, 19 Mar 2006 18:41:51 -0000, "Martin²" <never@give.one> wrote:
>Now you have to consider it as a quarterly rental :-(
Slightly cheaper than BT quarterly rental though Martin.
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| paul123 2006-03-21, 2:45 am |
|
Jono wrote:
>
> I buggered if I can get inbound to work with Asterisk.
>
Jono, I thought you were using AAH? I've eventually managed to get the
sipdiscount voip-in number working. I can either post the settings to
the group or email you them.
Paul
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| Sparks 2006-03-21, 2:45 am |
| "paul123" <paul@redy.net> wrote in message
news:1142796391.677780.152810@j33g2000cwa.googlegroups.com...
>
> Jono wrote:
>
>
> Jono, I thought you were using AAH? I've eventually managed to get the
> sipdiscount voip-in number working. I can either post the settings to
> the group or email you them.
>
> Paul
>
I would be interested in these too, so posting them for future reference
would be nice :-)
Sparks...
| |
|
| After serious thinking paul123 wrote :
> Jono wrote:
>
>
> Jono, I thought you were using AAH? I've eventually managed to get the
> sipdiscount voip-in number working. I can either post the settings to
> the group or email you them.
>
> Paul
Hi paul,
Yes it would be helpful if you could post the settings for AAH.
It's a bit of a pain ATM as my Sipdiscount number shows up as the CLI.
| |
| paul123 2006-03-21, 2:45 am |
|
Jono wrote:
> Yes it would be helpful if you could post the settings for AAH.
AAH Settings for Finarea voip-in numbers.
The following settings work for me and are the result of trying various
working Asterisk settings and adapting them to AAH. In the end, it was
a mix of 2 or 3 them. I'm sure there's a bit of "excess" and they could
be trimmed down. Oh and a plug.... Thanks to Asterisk@home! Great
prog/distro.
My settings are for voipstunt, but works for sipdiscount accounts too.
1:
In AMP - Setup - Trunks, create a SIP trunk
Dial Rules: {your dial rules}
Trunk name: voipstunt
PEER Details:
allow=ulaw&alaw
canreinvite=yes
context=custom-finareas
disallow=all
fromdomain=voipstunt.com
fromuser=USERNAME
host=sip.voipstunt.com
insecure=very
nat=yes
qualify=1000
secret=PASSWORD
srvlookup=yes
type=friend
username=USERNAME
USER Context: [empty]
USER Details: [empty]
Registration string: USERNAME:PASSWORD@sip.voipstunt.com/44208XXXXXXX
Submit and "red bar" it.
2:
In AMP - Setup - Inbound Routing, create a new entry
DID Number; 44208XXXXXXX
Set Destination: {your extension or ring group choice}
Submit and "red bar" it.
3:
In AMP - Maintenance - Config Edit
In extensions_custom.conf add:
[custom-finareas]
exten => USERNAME,1,Goto(ext-did,44208XXXXXXX,1)
submit and reread config
Notes:
My sip.conf general settings look like this (don't know how relevant
these are):
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
srvlookup=yes
callerid=Unknown
maxexpirey=180
defaultexpirey=160 ; Default length of incoming/outgoing registration
tos=reliability
progressinband=yes
context=from-pstn
sip_nat.conf contains:
externip=DYNDNSUSERNAME.dyndns.org
localnet=192.168.1.0/255.255.255.0
The inbound routing can be used with more than one finarea incoming
number. Simply create another trunk for the outgoing, an inbound DID
and an additional exten with your second voipstunt USERNAME2 and the
relevant number in the 44XXXXXXXXX format in the custom-finareas entry
in extensions_custom.conf.
Hope this works for you.
| |
|
| paul123 formulated on Sunday :
> Jono wrote:
>
> AAH Settings for Finarea voip-in numbers.
>
> The following settings work for me and are the result of trying various
> working Asterisk settings and adapting them to AAH. In the end, it was
> a mix of 2 or 3 them. I'm sure there's a bit of "excess" and they could
> be trimmed down. Oh and a plug.... Thanks to Asterisk@home! Great
> prog/distro.
>
> My settings are for voipstunt, but works for sipdiscount accounts too.
>
> 1:
> In AMP - Setup - Trunks, create a SIP trunk
>
> Dial Rules: {your dial rules}
> Trunk name: voipstunt
>
> PEER Details:
>
> allow=ulaw&alaw
> canreinvite=yes
> context=custom-finareas
> disallow=all
> fromdomain=voipstunt.com
> fromuser=USERNAME
> host=sip.voipstunt.com
> insecure=very
> nat=yes
> qualify=1000
> secret=PASSWORD
> srvlookup=yes
> type=friend
> username=USERNAME
>
> USER Context: [empty]
> USER Details: [empty]
>
> Registration string: USERNAME:PASSWORD@sip.voipstunt.com/44208XXXXXXX
>
> Submit and "red bar" it.
>
> 2:
> In AMP - Setup - Inbound Routing, create a new entry
>
> DID Number; 44208XXXXXXX
>
> Set Destination: {your extension or ring group choice}
>
> Submit and "red bar" it.
>
> 3:
> In AMP - Maintenance - Config Edit
>
> In extensions_custom.conf add:
>
> [custom-finareas]
> exten => USERNAME,1,Goto(ext-did,44208XXXXXXX,1)
>
> submit and reread config
>
> Notes:
> My sip.conf general settings look like this (don't know how relevant
> these are):
> [general]
> port = 5060 ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
> disallow=all
> allow=ulaw
> allow=alaw
> srvlookup=yes
> callerid=Unknown
> maxexpirey=180
> defaultexpirey=160 ; Default length of incoming/outgoing registration
> tos=reliability
> progressinband=yes
> context=from-pstn
>
>
> sip_nat.conf contains:
> externip=DYNDNSUSERNAME.dyndns.org
> localnet=192.168.1.0/255.255.255.0
>
> The inbound routing can be used with more than one finarea incoming
> number. Simply create another trunk for the outgoing, an inbound DID
> and an additional exten with your second voipstunt USERNAME2 and the
> relevant number in the 44XXXXXXXXX format in the custom-finareas entry
> in extensions_custom.conf.
>
> Hope this works for you.
Thanks for the info.
Unfortunately, I've got an 0560 number.......which, with the above
settings, I can only dial using Sipdiscount.
I have tried with BT, Orange, Sipgate, Voip.co.uk & Vodafone.
Think I'll open a new account & put an address where they provide
proper geo numbers.
Does anyone have a list of real area codes they (sipdiscount) allocate?
| |
| paul123 2006-03-21, 2:45 am |
| If there's a list available great, but I suspect it's a question of
"hit or miss" as the numbers seem to be allocated according to your
user details address. If they have numbers available from their pool
they give you a local geo, if not, you're probably stuck with an 0560.
You can certainly get 0208 and 01223 (Cambridge) numbers.
| |
|
| paul123 has brought this to us :
> If there's a list available great, but I suspect it's a question of
> "hit or miss" as the numbers seem to be allocated according to your
> user details address. If they have numbers available from their pool
> they give you a local geo, if not, you're probably stuck with an 0560.
>
> You can certainly get 0208 and 01223 (Cambridge) numbers.
Yes, you're right.
Anyway, I had a semi-dormant Internetcalls.com account where I'd not
added any user details.
I picked a random address in Birmingham & was rewarded with an 0121
number.
I have the same problem with this number - I can ring it fine from the
same trunk, but not from the "normal" networks.
| |
| M.Dexter@blueyonder.co.uk 2006-03-21, 2:45 am |
| On Sun, 19 Mar 2006 23:23:25 GMT, Jono <nothanks@notonyournelly.co.uk>
wrote:
>Think I'll open a new account & put an address where they provide
>proper geo numbers.
I thought of doing this Jono but I wonder if when you input your card
details somewhere in the system either their's or the banks the whole
thing will be cross checked ie card must match the address if you see
what I am getting at .
| |
|
| M.Dexter@blueyonder.co.uk formulated on Sunday :
> On Sun, 19 Mar 2006 23:23:25 GMT, Jono <nothanks@notonyournelly.co.uk>
> wrote:
>
>
> I thought of doing this Jono but I wonder if when you input your card
> details somewhere in the system either their's or the banks the whole
> thing will be cross checked ie card must match the address if you see
> what I am getting at .
Doesn't seem to be that slick.
You can pay before you enter your address (in my case, someone else's
address)
You have to enter "your" address before you "buy" a number.
Of course, I've corrected the personal details now I've got a proper
geo number - not that it's working properly, though.
| |
| Sparks 2006-03-21, 2:46 am |
|
"Jono" <nothanks@notonyournelly.co.uk> wrote in message
news:mn.9d7b7d633fce618e.48968@notonyournelly.co.uk...
> paul123 formulated on Sunday :
>
> Thanks for the info.
>
> Unfortunately, I've got an 0560 number.......which, with the above
> settings, I can only dial using Sipdiscount.
>
> I have tried with BT, Orange, Sipgate, Voip.co.uk & Vodafone.
>
> Think I'll open a new account & put an address where they provide proper
> geo numbers.
>
> Does anyone have a list of real area codes they (sipdiscount) allocate?
>
I have an 0560 number with VoIPCheap, and have just got it working thanks to
Paul!
I called it from my Orange mobile no problems (Except for hearing an
international type ringing tone!)
Calling from NTL gives me NU after 056015 - I will try to get NTL to add it!
I can try and call you if you like - I think you have my email?
Sparks....
| |
| {{{{{Welcome}}}}} 2006-03-21, 2:46 am |
| Thus spaketh Sparks:
>
> Just to also add, I just "bought" a SIP Discount number, it gave me a
> 0118 number, and this is also working - Thanks Paul!
>
> Sparks...
And also with an International ringing tone.
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| paul123 2006-03-21, 2:46 am |
| Glad it works Sparks! I was going bananas trying.
Jono, unless I did something else that's relevant that I didn't include
in the post, I dunno what to suggest..
| |
| Sparks 2006-03-21, 2:46 am |
|
"paul123" <paul@redy.net> wrote in message
news:1142815507.926856.81910@t31g2000cwb.googlegroups.com...
> Glad it works Sparks! I was going bananas trying.
>
> Jono, unless I did something else that's relevant that I didn't include
> in the post, I dunno what to suggest..
Hmm, maybe I spoke too soon!
If I leave it for a while, then try calling in, I get the following from AAH
(Below!)
But, if I make a call out from AAH on that trunk, it works.
It also sometimes just works randomly!?!
Any ideas!
Sparks...
-- Executing AbsoluteTimeout("SIP/5060-09a47df8", "15") in new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/5060-09a47df8", "") in new stack
== Spawn extension (from-sip-external, USERNAME, 2) exited non-zero on
'SIP/5060-09a47df8'
-- Executing AbsoluteTimeout("SIP/5060-09a47df8", "15") in new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/5060-09a47df8", "") in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/5060-09a47df8'
-- Executing AbsoluteTimeout("SIP/5060-09a47df8", "15") in new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/5060-09a47df8", "") in new stack
== Spawn extension (from-sip-external, USERNAME, 2) exited non-zero on
'SIP/5060-09a47df8'
-- Executing AbsoluteTimeout("SIP/5060-09a47df8", "15") in new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/5060-09a47df8", "") in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/5060-09a47df8'
| |
| paul123 2006-03-21, 2:46 am |
|
Sparks wrote:
> Hmm, maybe I spoke too soon!
>
> If I leave it for a while, then try calling in, I get the following from AAH
> (Below!)
> But, if I make a call out from AAH on that trunk, it works.
> It also sometimes just works randomly!?!
>
> Any ideas!
>
> Sparks...
>
>
> -- Executing AbsoluteTimeout("SIP/5060-09a47df8", "15") in new stack
> -- Set Absolute Timeout to 15
> -- Executing Congestion("SIP/5060-09a47df8", "") in new stack
> == Spawn extension (from-sip-external, USERNAME, 2) exited non-zero on
> 'SIP/5060-09a47df8'
> -- Executing AbsoluteTimeout("SIP/5060-09a47df8", "15") in new stack
> -- Set Absolute Timeout to 15
> -- Executing Congestion("SIP/5060-09a47df8", "") in new stack
> == Spawn extension (from-sip-external, h, 2) exited non-zero on
> 'SIP/5060-09a47df8'
> -- Executing AbsoluteTimeout("SIP/5060-09a47df8", "15") in new stack
> -- Set Absolute Timeout to 15
> -- Executing Congestion("SIP/5060-09a47df8", "") in new stack
> == Spawn extension (from-sip-external, USERNAME, 2) exited non-zero on
> 'SIP/5060-09a47df8'
> -- Executing AbsoluteTimeout("SIP/5060-09a47df8", "15") in new stack
> -- Set Absolute Timeout to 15
> -- Executing Congestion("SIP/5060-09a47df8", "") in new stack
> == Spawn extension (from-sip-external, h, 2) exited non-zero on
> 'SIP/5060-09a47df8'
I also had a problem with receiving calls - maybe 15 minutes or more
after rebooting AAH or rereading - it'd not receive calls to my voip-in
number. I'm not sure what I did to put it right (there was soooo much
fiddling!).
I'm no expert in this, but I suspect relevant settings for this problem
might be in the sip conf general settings with the expiry times. The
other thing I notice in your logs are the "from-sip-external" context.
I don't remember seeing that in my debug/log reports - just the
"ext-did" and "custom-finareas".
| |
|
| paul123 submitted this idea :
> Sparks wrote:
>
>
> I also had a problem with receiving calls - maybe 15 minutes or more
> after rebooting AAH or rereading - it'd not receive calls to my voip-in
> number. I'm not sure what I did to put it right (there was soooo much
> fiddling!).
>
> I'm no expert in this, but I suspect relevant settings for this problem
> might be in the sip conf general settings with the expiry times. The
> other thing I notice in your logs are the "from-sip-external" context.
> I don't remember seeing that in my debug/log reports - just the
> "ext-did" and "custom-finareas".
Update:
My internetcalls.com 0121 number /does/ work, sort of.
I have one-way audio on inbound & no ringing tone. DTMF passes OK
though.
| |
| paul123 2006-03-21, 2:46 am |
|
Jono wrote:
>
> Update:
>
> My internetcalls.com 0121 number /does/ work, sort of.
>
> I have one-way audio on inbound & no ringing tone. DTMF passes OK
> though.
Sounds like you've nearly cracked it.
Enzo Michelangeli's comments "Incoming call on VoIP-In issue" (about
halfway down the page) at http://www.voip-info.org/wiki/view/VoipStunt
might be relevant to your case.
or not...
| |
| My Name 2006-03-21, 7:46 am |
| On Sun, 19 Mar 2006 18:41:51 -0000, "Martin²" <never@give.one> wrote:
>Works pretty well, specially on VoIP / QoS router, occasionally need to
>re-dial, but that no bother.
>The only gripe I have that they now take your money after 120 days whether
>you used it up or not !
>Now you have to consider it as a quarterly rental :-(
>Regards,
>Martin
>
You can always top it up before the 120 days and get the period
extended by another 120 days. So the credit is still there and you can
always use it for calling mobiles if (like me) you only have a pay as
you go mobile or a distant cousin in Guatemala.
That can go on for a long time. After a year or two the credit is
still there, I assume, provided you have topped it up in time.
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