[asterisk-users] RE: zonedata.c
Jadrien Wauthier
jwauthier at dalcon.com
Sun May 13 19:41:33 MST 2007
Thank you very much. This helps a lot.
Jad
---------------
Date: Sun, 13 May 2007 10:58:44 +0300
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: Re: [asterisk-users] zonedata.c
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <20070513075844.GK3423 at xorcom.com>
Content-Type: text/plain; charset=us-ascii
On Sat, May 12, 2007 at 08:27:23PM -0500, Jadrien Wauthier wrote:
> Hi,
>
> Could anyone tell me how to read the values in the "zonedata.c" file?
> I am looking at the "zt_tone_ringtone" field mainly.
{ ZT_TONE_RINGTONE, "425/1000,0/4000" }
means a tone of frequency 425 Hz for 1000 ms and then silense (0 Hz) for
4000 ms.
{ ZT_TONE_RINGTONE, "400+450/400,0/200,400+450/400,0/2000" }
Similar notation. Though in this case the "frequency" is a bit more complex:
400+450 mean means a sound composed of a 400Hz and 450Hz frequencies.
(right?)
{ ZT_TONE_INFO, "!950/330,!1440/330,!1800/330,0/1000" }
The '!' tells that those sounds should not be repeated on the second
time this sound is played.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir at jabber.org
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
------------------------------
Message: 6
Date: Sun, 13 May 2007 10:40:37 +0200
From: Olivier <oza-4h07 at myamail.com>
Subject: Re: [asterisk-users] HPEC audio clipping
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<442fbb120705130140q305a6044xb7cd027fc2ad2751 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I didn't know that !
Thanks for the tip !
2007/5/11, Noah Miller <noahisaacmiller at gmail.com>:
>
> > Our last trial dates from 1.2.17 days (3 weeks ago).
> > My question is : are those HPEC audio clipping issues fixed with
> 1.2.17.1 ?
>
> It's not about the Asterisk version, it's about the HPEC version.
> According to other posts on the list, HPEC version 8.2 does not have
> the clipping issues, but the 9.x versions do.
>
> The echo "updates" to Asterisk 1.2.17.1 are to allow troubleshooting
> of the 9.x versions of HPEC so we can locate where the issues are.
> The updates do not fix the clipping issues themselves.
>
> If you have the clipping issue, make sure you get HPEC version 8.2 from
> Digium.
>
>
> - Noah
>
> > 2007/5/9, Matthew Fredrickson < creslin at digium.com>:
> > > If you contact Digium tech support directly they will provide you with
> > > the previous version of the echo canceler until the fix is made to the
> > > current version.
> > >
> > > Matthew Fredrickson
> > >
> > > On May 9, 2007, at 7:27 AM, Olivier wrote:
> > >
> > > > Any field return on this ?
> > > > Our last field trial of HPEC concluded we shouldn't use it at all,
> due
> > > > to audio clipping.
> > > >
> > > > Is it now fixed ?
> > > > _______________________________________________
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> > > >
> > > > asterisk-users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
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> > >
> >
> >
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> >
> >
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Message: 7
Date: Sun, 13 May 2007 19:21:06 +1000
From: Nick Adams <gmane at narkov.com>
Subject: [asterisk-users] Re: Snom 320 voicemail key & MWI
To: asterisk-users at lists.digium.com
Message-ID: <f26la2$2p3$1 at sea.gmane.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Stephen Bosch wrote:
> Ariel Monaco wrote:
>> Dear List,
>>
>> I'm having a blinking MWI light on the snom 320 even when there's no
>> message waiting in Asterisk.
>> We've managed to make the voicemail button work using
>> fromdomain=192.168.0.1 in sip.conf
>> vmexten=2500 (our VoicemailMain application extension in
>> extensions.conf). We also added
>> notifymimetype=application/simple-message-summary also in sip.conf to
>> allow SIP simple MWI
>> notifications.
>>
>> But the light is still blinking and there are no voicemail messages, any
>> ideas about how to address this
>> issue will be welcome.
>
> You've mentioned how your Asterisk server is configured, but how is the
> *phone* configured?
>
> If the MWI light on the phone is set to use the wrong mailbox, you would
> see a blinking light, even if you've erased all the messages in the
> mailbox that is accessed from the voicemail button.
>
> Two things are happening here:
>
> 1. You've got a button that you configure for retrieving messages
> 2. You've got a Message Waiting Indicator light that blinks when there
> are messages in the specified mailbox.
>
> Those are separate things -- you can have a button that retrieves from
> one box and a light that indicates messages in another box.
>
> Check your phone configuration again.
By default the Snom phones also use that light for missed calls.
------------------------------
Message: 8
Date: Sun, 13 May 2007 11:46:09 +0200
From: Per Jessen <per at computer.org>
Subject: Re: [asterisk-users] Asterisk High-Capacity Stability
To: asterisk-users at lists.digium.com
Message-ID: <f26mp2$tv1$1 at saturn.local.net>
Content-Type: text/plain; charset=utf-8
Atlanticnynex wrote:
> whether Asterisk could handle roughly one DS3's worth of calls (672
> calls) just doing the LCR (I've seen some pre-built LCR apps, looks
> like they all do on-the-fly MySQL queries- I think I'd write my own
> AGI that would use a cache).
When appropriately configured, MySQL does a pretty good job of caching
results too.
[129 lines snipped]
/Per Jessen, Zürich
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------------------------------
Message: 9
Date: Sun, 13 May 2007 13:00:27 +0100
From: "--[ UxBoD ]--" <uxbod at splatnix.net>
Subject: [asterisk-users] Zapateller and IAX2
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <20070513130027.4a5ae1e1 at uxbod.splatnix.net>
Content-Type: text/plain; charset=US-ASCII
Hi,
I have been using Zapateller with a TDM400 no problems at all, but
recently I have ported our BT number to a VoIP provider, and have a
strange problem. When I phone our number I first get the BT
unavailable three tone sound, and then it actually connects the call
via IAX2.
So, I disabled zapateller in the dialplan and tried again. Would you
believe it worked fine.
Has anybody else come across this ? I am using * 1.4.4.
Regards,
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------------------------------
Message: 10
Date: Sun, 13 May 2007 13:51:34 +0100
From: Steve Kennedy <steve-asterisk at gbnet.net>
Subject: Re: [asterisk-users] List of telemarketers??
To: asterisk-users at lists.digium.com
Message-ID: <20070513125134.GA25489 at colonelk.gbnet.net>
Content-Type: text/plain; charset=us-ascii
On Sun, May 13, 2007 at 01:54:08AM +0100, Chris Bagnall wrote:
> > 3. a list of bogus entries..so when you look at it, you know it's a
> > fake phone number...one that recently came in that got me thinking
> > this was 407 111 1111.
> I don't know much about the legal position over the other side of the pond, but I'm pretty sure that in the UK caller ID spoofing is illegal. There's nothing to stop you withholding your CLI of course, but to deliberately fake someone else's CLI (whether it exists or otherwise) pushes you over the line.
> Is the same not the case in the US?
I don't know if it's illegal (it would fall under the Comms Act if it
was), but I know it's discouraged. There are legal reasons you might
spoof CLI. Most telcos will have agreements that end-users can't do
nasty things with CLI (withholding doesn't actually block anything, just
flags the CLI should be withheld, so telcos, law enforcement etc still
get it).
Quite a few SS7 providers will allow customers to do what they like with
CLI, just have an agreement they wont do anything they shouldn't.
Steve
--
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------------------------------
Message: 11
Date: Sun, 13 May 2007 17:40:01 +0400
From: "Arun Kumar" <arunvoip at gmail.com>
Subject: [asterisk-users] TC400B load problem
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<a70a109b0705130640qcb0018do616ee33b2a9d676c at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi
Im trying to install my TC400B trans coder card when I do:
modprobe wctc4xxp
tail -f /var/log/messages says:
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with
92 transcoders (srcs=0000000c, dsts=00000101)
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with
92 transcoders (srcs=00000101, dsts=0000000c)
May 13 14:56:36 pbx2 kernel: Zaptel DTE (g.729a / g.723.1 5.3kbps)
Transcoder support LOADED (firm ver = 56)
May 13 14:56:49 pbx2 kernel: wctc4xxp: probe of 0000:03:01.0 failed with
error -5
please help
thanks
arun
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Message: 12
Date: Sun, 13 May 2007 10:11:09 -0400
From: Nitesh Divecha <nitesh at vipernetworks.com>
Subject: Re: [asterisk-users] Re: Snom 320 voicemail key & MWI
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <46471C7D.5040809 at vipernetworks.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Or you can specify "vmexten = *97" in sip.conf and your VM button will work.
Regards,
Nitesh
Nick Adams wrote:
> Stephen Bosch wrote:
>> Ariel Monaco wrote:
>>> Dear List,
>>>
>>> I'm having a blinking MWI light on the snom 320 even when there's no
>>> message waiting in Asterisk.
>>> We've managed to make the voicemail button work using
>>> fromdomain=192.168.0.1 in sip.conf
>>> vmexten=2500 (our VoicemailMain application extension in
>>> extensions.conf). We also added
>>> notifymimetype=application/simple-message-summary also in sip.conf to
>>> allow SIP simple MWI
>>> notifications.
>>>
>>> But the light is still blinking and there are no voicemail messages,
>>> any
>>> ideas about how to address this
>>> issue will be welcome.
>>
>> You've mentioned how your Asterisk server is configured, but how is the
>> *phone* configured?
>>
>> If the MWI light on the phone is set to use the wrong mailbox, you would
>> see a blinking light, even if you've erased all the messages in the
>> mailbox that is accessed from the voicemail button.
>>
>> Two things are happening here:
>>
>> 1. You've got a button that you configure for retrieving messages
>> 2. You've got a Message Waiting Indicator light that blinks when there
>> are messages in the specified mailbox.
>>
>> Those are separate things -- you can have a button that retrieves from
>> one box and a light that indicates messages in another box.
>>
>> Check your phone configuration again.
>
> By default the Snom phones also use that light for missed calls.
>
> _______________________________________________
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>
> asterisk-users mailing list
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
Message: 13
Date: Sun, 13 May 2007 12:50:41 -0400
From: Jon Pounder <JonP at inline.net>
Subject: Re: [asterisk-users] Dry Copper Pair
To: asterisk-users at lists.digium.com
Message-ID:
<20070513125041.1edo2nm8w0c00gw4 at webmail.relationalhost.com>
Content-Type: text/plain; charset=ISO-8859-1; DelSp="Yes";
format="flowed"
Quoting Stephen Bosch <posting at vodacomm.ca>:
> C F wrote:
>> Stephen i disagree. growing up in new work city i can say its quite
>> easy to get away with it in the city. where i live now in new jersey
>> (population of around 60000) i wouldnt be able to pull that off.
>
> The world is a big place, and I suppose there's room for all kinds. In
> these parts, the vigilance is pretty high. The pillars are padlocked
> now; they didn't use to be, and the COs are locked down like Fort Knox.
>
> Anyway, I know enough more than one person who has landed in the clink
> for treating the telco like a personal lab.
what exactly was the charge ?
- trespass - no its public land for the most part this stuff is on so
that doesn't apply
- vandalism/mischief - if no other customer was impacted I don't see
how this charge would stick since there is no measurable damages.
- theft of service ? Going rate for dry copper is under $20/month/pr
so to get up into the 5-10k level that might justify a higher level
theft charge with jail time that would take some time to add up.
Stealing cable TV/satellite probably works out to about 3x the monthly
rate of dry copper and I have never heard of anyone being told
anything more than disconnect it when they get caught.
I am not trying at all to justify the moral aspect of theft, I am just
making a point that I have never heard of anyone even getting in
trouble, let alone jail.
The other issue is what crime would be involved in assisting the telco
to deliver a better level of service by doing work yourself ?
For example I often do as much work on their side of the demarc as
possible when I have an order pending, then I know its done the way I
would have wanted it. I have never got anything other than a thank you
when the installer shows up and I just tell them where to make the
final connection.
Here is another what if - we had an adsl service in Mississauga at one
point that would never quite work properly, finally got the answer -
the line has a bridge tap on it somewhere but we won't remove it. WTF
? they contract to supply a service, now have an explanation why its
substandard yet won't fix it ? The point became moot as we cancelled
the service eventually, but say I had stuck my tone generator on it,
and walked back down the road to the CO and poked around till I found
the bridge tap and removed it.
What would they charge me with ? I'm only helping them fulfil a
contractural obligation they don't seem to want to meet. The way it
ended was they preferred to lose the account entirely rather than fix
a problem they knew about.
The other issue that hasn't even been touched in this thread is how
easy it is to just tap someone's line when everything is so exposed
like this. The tap might get found, but if it was a line powered radio
transmitter, chances of tracing back to the installer are minimal
unless someone saw it get installed.
>
> -Stephen-
> _______________________________________________
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Jon Pounder
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Message: 14
Date: Sun, 13 May 2007 20:54:55 +0300
From: "Dovid B" <asteriskusers at dovid.net>
Subject: Re: [asterisk-users] Double DTMF digits
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <009b01c79587$d316d140$0300a8c0 at DovidLaptop>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
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I am actually getting DTMF over SIP when people call in to a clients system
that is running a2billing. They are using RFC2833.
----- Original Message -----
From: "Remi Quezada" <remiq at monmouth.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Wednesday, May 09, 2007 6:15 PM
Subject: Re: [asterisk-users] Double DTMF digits
>I wonder if your hardware is doing the actual DTMF detecting. What
>hardware are you using? I'm using the TE205P and I believe that the DTMF
>detection is being done in the software in my case.
> Remi
>
> Steve Davies wrote:
>> On 5/3/07, Ken Leland III <k3leland at monmouth.com> wrote:
>>> When dtmfmode is set to inband for SIP, and i originate a call from sip
>>> out to the PSTN, I can hear the DTMF digit twice in the audio stream.
>>> Once very briefly and once for normal duration.
>>>
>>> Our Theory: While Asterisk is parsing the DTMF, for a fraction of a
>>> second, while the end user generated DTMF is being detected, the DTMF is
>>> passed inband. Once the DTMF is detected Asterisk silences it and
>>> regenerates it. Sensitive machines like auto attendants pick up both the
>>> brief end user generated tone as well as the full length asterisk
>>> generated tone and ultimately perceive each digit twice.
>>>
>>> Is anyone else experiencing this?
>>>
>>> I have reproduced this in an environment
>>> * with one asterisk server that is both the feature server and the
>>> media gateway, and is timing off of network T1s
>>> * with two servers, one feature server (timing off of ztdummy) and
>>> one media gateway (timing off of network T1s) using IAX as the inter
>>> asterisk protocol
>>>
>>> It is pretty easy to reproduce:
>>> -Dial a PSTN number(like your cell) from a sip phone using inband DTMF,
>>> and configured in asterisk sip.conf with dtmfmode=inband.
>>> -Answer the PSTN end.
>>> -Press and hold a digit on the sip phone. On the PSTN phone you will
>>> hear a very brief, end user generated, tone.
>>> -Let go of the digit on the sip phone. On the PSTN phone you will hear
>>> the asterisk generated tone.
>>>
>>> Can anyone else hear the brief initial tone? Any help is greatly
>>> appreciated!
>>
>> Yes, we have a similar issue, but do not normally use inband DTMF
>> because SIP phones very cleanly generate rfc2833 RTP packets directly
>> and remove this issue.
>>
>> On the other hand, asterisk is not alone dealing with this issue in
>> SIP. The Linksys ATAs have exactly the same issue.
>>
>> Strangely, I do not have a problem receiving inband DTMF through
>> Zaptel, which I believe uses the same DSP code for DTMF detection...
>> Or does it?
>>
>> Steve
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>>
>
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