Niccolò Belli | 1 Feb 2012 01:14
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Re: Proposed changes to Asterisk release and support cycles

I like it!

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Royce Souther | 1 Feb 2012 02:12
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Congestion outbound only with ATA boxes

I have an Asterisk server it runs great with SIP phones, soft SIP phones (twinkle) and a soft SIP phone app on my Android phone but I am having problems getting two ATA boxes working. I have a Linksys PAP2T, it is unlocked and I have used them before with no problems. I was able to receive calls with from any local SIP phone or from my Link2VoIP connection via the Internet but it could not call out. It could not call out to the Link2VoIP or any of the SIP phones. I spent a lot of time going over the configureation for this Asterisk server and the settings in the Linksys PAP2T box but could not get it to work. I removed the Linksys PAP2T and replaced it with an HT503 because I read a lot of good recommendations for this device. It seems to have almost the same problem. I say almost because when the Linksys would get congestion I would hear the Asterisk recording tell me "All circuits are busy now, good-bye" but the HT503 only gets a busy tone.

All the SIP phones can call out no problem but these two ATA boxes that I am trying to use the FXS ports to connect old analog POTS phones to are not working.

I have turned on the debug in Asterisk and can see the point where I get congestion but I don't know how to make Asterisk give me more details as to why I am getting congestion. Can anyone help me to get more details about this problem?

I traced the debug from a working SIP phone as it makes an outgoing call and from the HT503 as it tries to make a call. Everything is identical right up to the point where the HT503 gets a congestion instruction from the Asterisk server.
Here is the debug output just at the point where it happens.

    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s <at> macro-dial:7] Dial("SIP/302-08221a38", "SIP/301||tr") in new stack
    -- Called 301
Home*CLI>
<--- Transmitting (NAT) to 192.168.0.100:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK443855200;received=192.168.0.100;rport=5060
From: <sip:302 <at> 192.168.0.1>;tag=1257222779
To: <sip:301 <at> 192.168.0.1>;tag=as201c8013
Call-ID: 979693319-5060-5 <at> 192.168.0.100
CSeq: 41 INVITE
User-Agent: FPBX-2.4.0(1.4.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301 <at> 192.168.0.1>
Content-Length: 0


<------------>
    -- SIP/301-0822de30 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s <at> macro-dial:8] Set("SIP/302-08221a38", "DIALSTATUS=CONGESTION") in new stack
    -- Executing [s <at> macro-exten-vm:10] Set("SIP/302-08221a38", "SV_DIALSTATUS=CONGESTION") in new stack
    -- Executing [s <at> macro-exten-vm:11] GosubIf("SIP/302-08221a38", "0?docfu|1") in new stack
    -- Executing [s <at> macro-exten-vm:12] GosubIf("SIP/302-08221a38", "0?docfb|1") in new stack
    -- Executing [s <at> macro-exten-vm:13] Set("SIP/302-08221a38", "DIALSTATUS=CONGESTION") in new stack
    -- Executing [s <at> macro-exten-vm:14] NoOp("SIP/302-08221a38", "Voicemail is novm") in new stack
    -- Executing [s <at> macro-exten-vm:15] GotoIf("SIP/302-08221a38", "1?s-CONGESTION|1") in new stack
    -- Goto (macro-exten-vm,s-CONGESTION,1)
    -- Executing [s-CONGESTION <at> macro-exten-vm:1] PlayTones("SIP/302-08221a38", "congestion") in new stack
Audio is at 192.168.0.1 port 10162
Adding codec 0x100 (g729) to SDP

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Hans Witvliet | 1 Feb 2012 09:25
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Re: Proposed changes to Asterisk release and support cycles

On Tue, 2012-01-31 at 15:52 -0500, John Knight wrote:
> > I like the idea of LTR release more often that would have the
> > feature patches baked in.  Case in point the new conference app
> > requires a jump to version 10 while the 1.8 conference app is quite
> > useless but 1.8 is my LTR version so I am stuck without the
> > conference app in my mainline systems for two years. 
> 
> Well said!  This is also true of any type of long term supported
> release whether if it's an operating system, application, etc.  In the
> "LTS" name, it conjurs up thoughts of Ubuntu, but comparisons to
> RHEL/Fedora are far more appropriate I would think as Ubuntu focuses
> nearly exclusively on new point releases while backporting new
> features is what a company like Red Hat excels at and should be the
> prime example of how to run dual software channels (enterprise release
> in RHEL vs. hobby release in Fedora). 
> 
....

> 
> I know distros and applications are two fundamentally different
> things, with entirely different goals and requirements, but I still
> think Red Hat provides the best example because 1) they have turned it
> into a science how smooth their development process goes in ratio to
> satisfied customers and 2) it's the only other open source software
> project I can think of that can accurately compare.  In a past meeting
> I had with Digium while working for another company, they too directly
> drew a correlation between the new LTS idea and ubuntu lts/non-lts and
> rhel/fedora.
> 
> The conference app changes since 1.4 I haven't been thrilled with, but
> in the whole time I've been supporting 1.8.x for my customers, I've
> come up with a very stable solution building on it and I haven't had
> any surprises come my way.   
> 
very well said indeed.
Some (...) distro's think dat LTS implies a complete feature freeze.
Others are more flexibel about it, that besides current versions of
applications, they are willing to support both elder _and_ newer
versions. (as example, i'm refering to the fact that hours after the
anouncement, firefox10 became available for sles11)

As said, re-written features like conference, are that important that
one shouldn't have to wait years for the next LTS. So this overlap of
multiple LTS-versions looks very much attractive

Having said that, i do understand that multiple versions of
features/applications puts an huge extra burden on the people who have
to maintain both versions, as the original version (as the term LTS
implies) should be maintained with all its limitations also.

hw

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Daniel Pocock | 1 Feb 2012 09:36
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Re: CA Issued Certificates / TLS + SRTP


>>>>> * And, is it necessary to use both my server specific certificate and
>>>>> the intermediate certificate on the telephones or will the telephones
>>>>> only require the server specific certificate?
>>>> The phones should already have the root certificate for Geotrust, you
>>>> should not deploy intermediate roots into the phones if you can
>>>> avoid it
>>> If I understand this correctly (and the other emails you sent), the
>>> Polycom does not need any preloaded certificates / keys, it will ask the
>>> CA and then evaluate the certificate provided by Asterisk during TLS
>>> setup; is that correct? Makes it much easier. (Unfortunately my Polycom
>>> is a bit old so I will have to see if I can upgrade it.)

By `preloaded', I mean you should not have to load any certificates or
key pairs manually into the phones

The phones should have the default CA certs that come in the firmware

Most recent Polycom phones also have a client certificate and private
key built in.  This allows you to secure the provisioning process.

Some of the older Polycoms went end-of-life, some don't have client
certs built in, so you'll have to research all that carefully on their
support site.  E.g. the IP 300, IP 430 and IP 500 are too old for proper
TLS, while the IP321, IP 450 and IP550 are good

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Stuart Elvish | 1 Feb 2012 10:58
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Re: CA Issued Certificates / TLS + SRTP

Thanks for the clarification. I have looked at Polycom's website and
saw which phones have the latest firmware (or at least a firmware that
supports TLS) available.

Didn't get around to the testing with the chained certificate but will
try again this evening.

>
>>>>>> * And, is it necessary to use both my server specific certificate and
>>>>>> the intermediate certificate on the telephones or will the telephones
>>>>>> only require the server specific certificate?
>>>>> The phones should already have the root certificate for Geotrust, you
>>>>> should not deploy intermediate roots into the phones if you can
>>>>> avoid it
>>>> If I understand this correctly (and the other emails you sent), the
>>>> Polycom does not need any preloaded certificates / keys, it will ask the
>>>> CA and then evaluate the certificate provided by Asterisk during TLS
>>>> setup; is that correct? Makes it much easier. (Unfortunately my Polycom
>>>> is a bit old so I will have to see if I can upgrade it.)
>
>
>
> By `preloaded', I mean you should not have to load any certificates or
> key pairs manually into the phones
>
> The phones should have the default CA certs that come in the firmware
>
> Most recent Polycom phones also have a client certificate and private
> key built in.  This allows you to secure the provisioning process.
>
> Some of the older Polycoms went end-of-life, some don't have client
> certs built in, so you'll have to research all that carefully on their
> support site.  E.g. the IP 300, IP 430 and IP 500 are too old for proper
> TLS, while the IP321, IP 450 and IP550 are good
>
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Kingsley Tart | 1 Feb 2012 11:34
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read digits during recording / DTMF in conference?

Hi,

I want to create a system for incoming calls where, under some
circumstances, callers get routed straight to voicemail (or some other
means of recording a message) but if they enter a valid extension number
then the recorded message would be abandoned and they'd be diverted to
the extension number they entered.

I realise this can be done with the voicemail app with operator=yes but
the problem with this is that the caller has to press 0 while the
announcement is being played. If they're too slow and recording has
started, they've missed the opportunity.

So I played around with ConfBridge and a couple of call files, just to
see if I could get it to work. It's a bit convoluted but the idea is
that the caller gets silently put into a conference, then two call files
make asterisk silently connect to other calls into the same conference,
with one doing the recording and the other using Read() to collect
digits.

If I just had the caller and one of the other calls in the conference
(the one doing Read()) then this worked - Read() managed to read the
DTMF digits and assign them to a variable.

However, when the 'recording' call is also in the conference, the 'read'
call can no longer recognise the DTMF digits. To test, I made the 'read'
call play a sound before calling Read() and I could hear this being
played so the call was definitely there. However, regardless of the
number of digits I pressed, Read() didn't notice any of them, even if I
introduced a delay so that the other channels were quiet before the call
to Read().

I realise this might seem a bit like a mad solution but can anyone else
think of a way to get Asterisk to read (and react to) DTMF digits during
a recording?

This is with Asterisk 1.8.7.

--

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Cheers,
Kingsley.

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isrlgb | 1 Feb 2012 11:45
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Re: read digits during recording / DTMF in conference?

M…
-----Original Message-----
From: Kingsley Tart <kingsley <at> skymarket.co.uk>
Sender: asterisk-users-bounces <at> lists.digium.com
Date: Wed, 01 Feb 2012 10:34:07 
To: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users <at> lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users <at> lists.digium.com>
Subject: [asterisk-users] read digits during recording / DTMF in conference?

Hi,

I want to create a system for incoming calls where, under some
circumstances, callers get routed straight to voicemail (or some other
means of recording a message) but if they enter a valid extension number
then the recorded message would be abandoned and they'd be diverted to
the extension number they entered.

I realise this can be done with the voicemail app with operator=yes but
the problem with this is that the caller has to press 0 while the
announcement is being played. If they're too slow and recording has
started, they've missed the opportunity.

So I played around with ConfBridge and a couple of call files, just to
see if I could get it to work. It's a bit convoluted but the idea is
that the caller gets silently put into a conference, then two call files
make asterisk silently connect to other calls into the same conference,
with one doing the recording and the other using Read() to collect
digits.

If I just had the caller and one of the other calls in the conference
(the one doing Read()) then this worked - Read() managed to read the
DTMF digits and assign them to a variable.

However, when the 'recording' call is also in the conference, the 'read'
call can no longer recognise the DTMF digits. To test, I made the 'read'
call play a sound before calling Read() and I could hear this being
played so the call was definitely there. However, regardless of the
number of digits I pressed, Read() didn't notice any of them, even if I
introduced a delay so that the other channels were quiet before the call
to Read().

I realise this might seem a bit like a mad solution but can anyone else
think of a way to get Asterisk to read (and react to) DTMF digits during
a recording?

This is with Asterisk 1.8.7.

-- 
Cheers,
Kingsley.

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Andrew Nowrot | 1 Feb 2012 12:38
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Asterisk 10.0 Realtime

Hi

I have noticed new behaviour of asterisk 10.0 realtime.
In 1.6 when I was using realtime:

"""
[somecontext]

 exten => someexten1......
 exten => someexten2......
 exten => someexten3......
 exten => someexten4......

switch => Realtime/${CONTEXT} <at> extensions
"""

switch statement was executed after lines above (so there was a
precedence of the lines declared in a extensions.conf over the ones in
database).

In asterisk 10.0 switch is executed before extens declared in the
extensions.conf file.

Is there a way to change that and have previous behaviour?

Cheers

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Daniel Pocock | 1 Feb 2012 13:06
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Re: CA Issued Certificates / TLS + SRTP


On 01/02/12 10:58, Stuart Elvish wrote:
> Thanks for the clarification. I have looked at Polycom's website and
> saw which phones have the latest firmware (or at least a firmware that
> supports TLS) available.
> 
> Didn't get around to the testing with the chained certificate but will
> try again this evening.
> 
> 

One thing that frustrates people about Polycom is the very limited list
of root CAs they support - it was probably OK when they first started
doing SSL, but things have changed a lot now

The latest phones (e.g. IP321) have more memory than those they replace
(e.g. IP320) and so they should be able to handle a larger list of built
in root CAs (which Polycom can distribute through the firmware update).
 The obvious ones that are missing are the budget CAs:

- CaCert.org (all certs are free)
- startssl.com  (which has some free certs)
- GoDaddy

These budget CAs are now supported by the various Linux distributions
and Android phones, so they are clearly above a certain threshold of
stability

Polycom phones should also be able to handle 4096 bit certs with the
extra memory, but that appears to need remediation in the firmware (I
tried installing a custom 4096 bit cert and it didn't accept it)

If anyone is registered with Polycom as a reseller, they can quote these
issue numbers:

EXT-3192 GoDaddy root CA cert
https://jira.polycom.com:8443/browse/EXT-3192

EXT-3193 CACert root CA cert
https://jira.polycom.com:8443/browse/EXT-3193

EXT-3238 Support for 4096 bit keys
https://jira.polycom.com:8443/browse/EXT-3238

As in most commercial enterprises, the more customers who request fixes
on these issues, the higher it will go on their priority list

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Christian Gansberger | 1 Feb 2012 13:29
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SIP Provider Russia, Ukraine, Poland

Hello List!

I'm searching for SIP-Providers in the following countries:
Russia
Ukraine
Poland

I need a geographical number for each country, maybe a prepaid
SIP-Account, trunking is not important.
Has anyone some experience with these countries?

yours
christian

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Gmane