This works perfectly with the exception of not being able to send DTMF
tones over the line - dialling is no problem. Therefore I'm unable to
call most business lines which answer with "For xxxx please press y".
This also rules out voicemail.
I've changed the analogue handset to an alternative which makes no
difference. The ATA186 is running Version: v3.1.0 atasip (Build
040211A).
> This works perfectly with the exception of not being able to send DTMF
> tones over the line - dialling is no problem. Therefore I'm unable to
> call most business lines which answer with "For xxxx please press y".
> This also rules out voicemail.
> Any ideas, please?
are there any options to configure "in band signalling " or similar
DTMF options on the ATA ?
In message <1157451905.739407.67290@i42g2000cwa.googlegroups. com>, PhilT
<newsnet@gmail.com> writes
>
>Malcolm Loades wrote:
>
>> This works perfectly with the exception of not being able to send DTMF
>> tones over the line - dialling is no problem. Therefore I'm unable to
>> call most business lines which answer with "For xxxx please press y".
>> This also rules out voicemail.
>
>> Any ideas, please?
>
>are there any options to configure "in band signalling " or similar
>DTMF options on the ATA ?
Not that I can see.
Sipgate have very detailed instructions for setting up the ATA186 on
their website including a screen shot of the ATA186 configuration page
(only accesible by registered users) and mine appears to match theirs.
I asked Sipgate support for help but they asked the same question as
you! I assume the ATA186 should work on Sipgate, otherwise why have
such detailed configuration details for it? Sipgate ignore this
question :-(
Does anyone here successfully use an ATA186 and Sipgate? Does anyone
use an ATA186 successfully with another provider?
"Malcolm Loades" <devnull@spam-trap.org.uk> wrote in
message news:+E3lp0KBZW$EFwUz@news.posting
> In message
> <1157451905.739407.67290@i42g2000cwa.googlegroups. com>,
> PhilT <newsnet@gmail.com> writes
> >
> > Malcolm Loades wrote:
> >
> > > This works perfectly with the exception of not being
> > > able to send DTMF tones over the line - dialling is
> > > no problem. Therefore I'm unable to call most
> > > business lines which answer with "For xxxx please
> > > press y". This also rules out voicemail.
> >
> > > Any ideas, please?
> >
> > are there any options to configure "in band signalling
> > " or similar DTMF options on the ATA ?
>
> Not that I can see.
There should be an option somewhere, I've not seen a 186 however the 486 I
had was configurable in this way, I can't remember the details offhand.
However, as far as Sipgate voicemail goes, if you dial the remote access
number 020 7043 7777 then you should be able to access it ok, let me know
if you have problems. Note that this number will ask you for your SIP ID
and 4-digit PIN.
Malcolm Loades wrote:
> This works perfectly with the exception of not being able to send DTMF
> tones over the line - dialling is no problem. Therefore I'm unable to
> call most business lines which answer with "For xxxx please press y".
> This also rules out voicemail.
The same problem was reported to me yesterday but with sipgate and an
ATA1100.
I try and find out whether they got it working or not.
In message <4m5olfF4l8mbU1@individual.net>, Ivor Jones
<ivor@despammed.invalid> writes
>"Malcolm Loades" <devnull@spam-trap.org.uk> wrote in
>message news:+E3lp0KBZW$EFwUz@news.posting
>> In message
>> <1157451905.739407.67290@i42g2000cwa.googlegroups. com>,
>> PhilT <newsnet@gmail.com> writes
>> >
>> > Malcolm Loades wrote:
>> >
>> > > This works perfectly with the exception of not being
>> > > able to send DTMF tones over the line - dialling is
>> > > no problem. Therefore I'm unable to call most
>> > > business lines which answer with "For xxxx please
>> > > press y". This also rules out voicemail.
>> >
>> > > Any ideas, please?
>> >
>> > are there any options to configure "in band signalling
>> > " or similar DTMF options on the ATA ?
>>
>> Not that I can see.
>
>There should be an option somewhere, I've not seen a 186 however the 486 I
>had was configurable in this way, I can't remember the details offhand.
>
>However, as far as Sipgate voicemail goes, if you dial the remote access
>number 020 7043 7777 then you should be able to access it ok, let me know
>if you have problems. Note that this number will ask you for your SIP ID
>and 4-digit PIN.
Fine if done from a 'regular' phone, but no good using the
ATA186/Sipgate combination because the tones aren't recognised!
In message <1157475174.643877.48770@d34g2000cwd.googlegroups. com>, PhilT
<newsnet@gmail.com> writes
>Malcolm Loades wrote:
>
>> Does anyone here successfully use an ATA186 and Sipgate? Does anyone
>> use an ATA186 successfully with another provider?
>>
>
>I have used mine with FWD as well as BT. Its resting at the moment.
>(Assuming we're talking about a Cisco ATA186).
>
>http://www.geckotechllc.com/pdf/CiscoATA18xguide.pdf may help. There's
>a reference on p94 to an audio settings bitmap :-
are you saying the audio bitmap does not exist on the SIP config ?
Having run both MGCP and SIP there is a lot of common ground on the
settings, as is clear from the Cisco documentation on the web.
The Sipgate page shows "0x00150015" as the "AudioMode:" setting.
The default in the manual is 0x00350035
With my limited bitmap skills it appears that the Sipgate setting is
for "by negotiation" so I suggest you try 0x00050005 which is "DTMF
always in band".
"Malcolm Loades" <devnull@spam-trap.org.uk> wrote in
message news:9OJFLjM5$d$EFwla@news.posting
> In message <4m5olfF4l8mbU1@individual.net>, Ivor Jones
> <ivor@despammed.invalid> writes
[snip]
> > However, as far as Sipgate voicemail goes, if you dial
> > the remote access number 020 7043 7777 then you should
> > be able to access it ok, let me know if you have
> > problems. Note that this number will ask you for your
> > SIP ID and 4-digit PIN.
>
> Fine if done from a 'regular' phone, but no good using the
> ATA186/Sipgate combination because the tones aren't
> recognised!
Ah, I see what you mean. Can you send me a screenshot of the configuration
page..? Send to g6urp01 at googlemail dot com and I'll have a look and see
if I can see anything obviously out of place.
PhilT wrote:
> Malcolm Loades wrote:
>
>> Unfortunately this guide is for MGCP not SIP.
>
> are you saying the audio bitmap does not exist on the SIP config ?
>
> Having run both MGCP and SIP there is a lot of common ground on the
> settings, as is clear from the Cisco documentation on the web.
>
> The Sipgate page shows "0x00150015" as the "AudioMode:" setting.
>
> The default in the manual is 0x00350035
>
> With my limited bitmap skills it appears that the Sipgate setting is
> for "by negotiation" so I suggest you try 0x00050005 which is "DTMF
> always in band".
>
Of course you will want this to be always outband if you use G.729.
so presumably 0x00250025.
Malcolm Loades wrote:
> This works perfectly with the exception of not being able to send DTMF
> tones over the line - dialling is no problem. Therefore I'm unable to
> call most business lines which answer with "For xxxx please press y".
> This also rules out voicemail.
>
> I've changed the analogue handset to an alternative which makes no
> difference. The ATA186 is running Version: v3.1.0 atasip (Build
> 040211A).
>
> Any ideas, please?
This sounds exactly the same as the problem I have using the
SMC7908VoWBRA router. DTMF tones are not recognized so I cannot use the
Sipgate voicemail by dialling 50000. A few months ago I spent quite a
lot of time emailing Sipgate tech support without resolving it. I seem
to recall discovering that there are two varieties of out of band DTMF
- perhaps my SMC and your device only implements one of them.
I tried again today as I found out that there is a firmware upgrade for
my device, but still no luck unfortunately. It's annoying, as it works
great otherwise.
PhilT wrote:
> Thomas Kenyon wrote:
>
>> Of course you will want this to be always outband if you use G.729.
>> so presumably 0x00250025.
>
> depends whether outband works or not I suppose :-)
>
> Phil
>
True, but inband isn't supposed to work with that codec. (well, support
is non-standard).
In message <1157491110.627420.121870@p79g2000cwp.googlegroups .com>,
PhilT <newsnet@gmail.com> writes
>
>Malcolm Loades wrote:
>
>> Unfortunately this guide is for MGCP not SIP.
>
>are you saying the audio bitmap does not exist on the SIP config ?
>
>Having run both MGCP and SIP there is a lot of common ground on the
>settings, as is clear from the Cisco documentation on the web.
>
>The Sipgate page shows "0x00150015" as the "AudioMode:" setting.
>
>The default in the manual is 0x00350035
>
>With my limited bitmap skills it appears that the Sipgate setting is
>for "by negotiation" so I suggest you try 0x00050005 which is "DTMF
>always in band".
>
Thanks for the suggestions, and also everyone else who responded. My
setting was 0x00150015 as per the Sipgate page. I've tried 0 and 2
...... still no DTMF detected by the other end.
More Googling brings up lots of reports similar to mine but no solution
:-( Unless someone knows better!
Malcolm Loades wrote:
> In message <1157491110.627420.121870@p79g2000cwp.googlegroups .com>,
> PhilT <newsnet@gmail.com> writes
> >The Sipgate page shows "0x00150015" as the "AudioMode:" setting.
> >
> >The default in the manual is 0x00350035
> >
> >With my limited bitmap skills it appears that the Sipgate setting is
> >for "by negotiation" so I suggest you try 0x00050005 which is "DTMF
> >always in band".
> >
> Thanks for the suggestions, and also everyone else who responded. My
> setting was 0x00150015 as per the Sipgate page. I've tried 0 and 2
> ..... still no DTMF detected by the other end.
>
0x00050005 works fine with sipgate on an ATA186 calling out to a BT
line, I can hear the tones and they work with Call Minder's IVR menu
including accepting a PIN and navigating the menu. They don't seem to
work with Sipgate's voicemail.
In message <1157895891.437427.317100@d34g2000cwd.googlegroups .com>,
PhilT <newsnet@gmail.com> writes
>
>Malcolm Loades wrote:
>> In message <1157491110.627420.121870@p79g2000cwp.googlegroups .com>,
>> PhilT <newsnet@gmail.com> writes
>> >The Sipgate page shows "0x00150015" as the "AudioMode:" setting.
>> >
>> >The default in the manual is 0x00350035
>> >
>> >With my limited bitmap skills it appears that the Sipgate setting is
>> >for "by negotiation" so I suggest you try 0x00050005 which is "DTMF
>> >always in band".
>> >
>> Thanks for the suggestions, and also everyone else who responded. My
>> setting was 0x00150015 as per the Sipgate page. I've tried 0 and 2
>> ..... still no DTMF detected by the other end.
>>
>
>0x00050005 works fine with sipgate on an ATA186 calling out to a BT
>line, I can hear the tones and they work with Call Minder's IVR menu
>including accepting a PIN and navigating the menu. They don't seem to
>work with Sipgate's voicemail.
>
>Version: v3.1.0 atasip (Build 040211A)
>
Superb! Thanks!
I'd been doing the testing of each setting by calling Sipgate's
voicemail - no wonder it failed. The setting you suggest does indeed
work fine otherwise - I can live with that.
"PhilT" <newsnet@gmail.com> wrote in message
news:1157994927.745878.126390@h48g2000cwc.googlegr oups.com
> Malcolm Loades wrote:
>
> > I'd been doing the testing of each setting by calling
> > Sipgate's voicemail - no wonder it failed.
>
> I'm afraid I have learned that...
>
> testing + sipgate = insanity + invalid conclusions
>
> the variability of the service is too great for robust
> testing.
How so..? It works fine here and has done almost 100% since the server
upgrades in April.
Ivor Jones wrote:
> "PhilT" <newsnet@gmail.com> wrote in message
> news:1157994927.745878.126390@h48g2000cwc.googlegr oups.com
>
> > the variability of the service is too great for robust
> > testing.
>
> How so..? It works fine here and has done almost 100% since the server
> upgrades in April.
"almost 100%" may be the problem, I am looking for something for
"robust testing". You say yourself that you don't make many calls, so
maybe your sample is statistically challenged.
My reality was that having configured an ATA to use Sipgate it failed
to register for a good while, so I started messing with settings and
poking the firewall looking for solutions to the problem in order to
get a dial tone. Meantime I setup logging that showed the ATA trying to
register and failing - did I have the wrong password ? After a while I
took a break from it and came back later to discover that it had
registered, so the succession of failures followed by a success were
due to something other than my settings - the server wasn't responding.
Having got it up and running some of the calls to BT lines had no ring
tone, note *some* of the calls. Hardly ideal if trying to demonstrate
it to someone new to VoIP "this isn't ringing" "it is but there isn't a
ring tone this time" "yeah, right".
Later on I noticed the ATA had disappeared from the status indicator,
presumably a registration failure again.
I haven't managed to dial into the geo number and have the ATA respond,
I will go and test this on another known working setup first in order
to be sure that the service is working properly - once bitten twice
shy.
I set a friend up with Sipgate several months ago, she is on a wireless
network and the service sometimes works and sometimes doesn't. As the
network and the ATA settings are constant through this I am drawn to
the conclusion that Sipgate is the variable element.
Another guy I know has an asterisk server with a sipgate account on it
and he regularly sees authentication failures and other issues with
servers not responding.
So I conclude that if something doesn't work and Sipgate is the
provider then there's a greater than 50/50 chance that Sipgate is the
problem and not the end user's setup.
PhilT wrote:
> Ivor Jones wrote:
>> "PhilT" <newsnet@gmail.com> wrote in message
>> news:1157994927.745878.126390@h48g2000cwc.googlegr oups.com
>>
>>> the variability of the service is too great for robust
>>> testing.
>> How so..? It works fine here and has done almost 100% since the server
>> upgrades in April.
>
> "almost 100%" may be the problem, I am looking for something for
> "robust testing". You say yourself that you don't make many calls, so
> maybe your sample is statistically challenged.
>
Even Gradwell doesn't work 100% of the time. (or BT for that matter).
Sipgate and ATA186 
Linear Mode
