[asterisk-users] VoIP support engineer opportunity

[David Cunningham]

Hello,

Voisonics is hiring a VoIP support engineer to assist our customers running
Asterisk based hosted PBX platforms. This is a part-time contract
work-from-home position.

For communication reasons we're looking for someone in a timezone
encompassing New Zealand, Canada, the USA, and Mexico. If you are not
physically located in that area please do not apply - being "flexible" from
another part of the world is *not* what we're looking for.

The role involves providing technical support of Asterisk based PBX
platforms to our customer's technical staff, Linux system administration,
and small dev-ops type development projects. It does not involve providing
technical support to end users or the general public.

Customers are located around the world. You will generally be responding
during your business hours, though sometimes out of hours work will be
necessary. Once training is completed, the position will involve providing
24x7 on-call emergency cover in rotation with other staff.

Must-have:
1. Fluent command-line Linux ability on Ubuntu, Debian, Rocky, CentOS,
and/or RedHat.
2. Asterisk administration and configuration experience.
3. SIP debugging experience. For example, you should know what packets are
typically involved in setting up a call.
4. Experience with administration and configuration of Apache and MySQL or
PostgreSQL.
5. Ability to program in Perl or shell scripts.
6. Good written and verbal English language ability.
7. Ability to solve problems and create solutions independently. Although
there are other staff, most work is done by one engineer on their own.
8. Have experience providing professional IT support to business.
9. Be an individual self-employed contractor.

Nice-to-have:
1. Experience with Kamailio, NFS, GlusterFS, Puppet, or Zabbix.
2. C, Go, Javascript, AngularJS, HTML or CSS programming ability.
3. Advanced network knowledge (beyond basic Linux networking which is a
must-have).

To apply for this role please email me off-list. Do not call by phone
unless arranged in advance.

In your application:
1. List your experience/compatibility for each of the must-have
requirements individually, plus any of the nice-to-have items you fit as
well.
2. Provide your physical location, hours of availability, and indication of
hourly rate.
3. Let us know what other work you already have during your hours of
availability.
4. A full CV is welcome.

Thank you,

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
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Re: [asterisk-users] RTP address learning and timing problem

[David Cunningham]

Thank you Joshua.


On Tue, 18 Apr 2023 at 21:18, Joshua C. Colp  wrote:

> I don't know in that specific output what happened. Your best course of
> action is to add further logging or step through the logic with all of the
> knowledge you have of the RTP streams to understand what is happening.
>
> On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Joshua,
>>
>> Thank you for that. From the code it kind of looks like
>> STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum:
>>
>> if (!ast_sockaddr_isnull(>strict_rtp_address)
>> && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(),
>> rtp->rtp_source_learn.start)) {
>> ast_verb(4, "%p -- Strict RTP learning complete - Locking on source
>> address %s\n",
>>
>> Our call shows:
>>
>> # grep C-00024cd5 full.log | egrep 'Strict RTP'
>> [Feb 22 11:16:41] VERBOSE[29023][C-00024cd5] res_rtp_asterisk.c:>
>> 0x2b308c074f80 -- Strict RTP learning after remote address set to:
>> xx.xx.154.111:18578
>> [Feb 22 11:17:00] VERBOSE[29023][C-00024cd5] res_rtp_asterisk.c:>
>> 0x2b315c01cbc0 -- Strict RTP learning after remote address set to:
>> xx.xx.0.12:16498
>> [Feb 22 11:17:00] VERBOSE[28191][C-00024cd5] res_rtp_asterisk.c:>
>> 0x2b308c074f80 -- Strict RTP switching to RTP remote address
>> xx.xx.154.111:18578 as source
>> [Feb 22 11:17:00] VERBOSE[28191][C-00024cd5] res_rtp_asterisk.c:>
>> 0x2b308c074f80 -- Strict RTP learning complete - Locking on source address
>> xx.xx.154.111:18578
>> [Feb 22 11:17:00] VERBOSE[28194][C-00024cd5] res_rtp_asterisk.c:>
>> 0x2b315c01cbc0 -- Strict RTP switching source address to xx.xx.114.237:16498
>> [Feb 22 11:17:01] VERBOSE[28194][C-00024cd5] res_rtp_asterisk.c:>
>> 0x2b315c01cbc0 -- Strict RTP learning complete - Locking on source address
>> xx.xx.114.237:16498
>>
>> I'm a bit confused because the second "Strict RTP learning after remote
>> address set" should reset the rtp_source_learn.start timestamp, and yet the
>> "Strict RTP learning complete" messages are less than 5000ms after that.
>> What could be happening?
>>
>> Thanks again.
>>
>>
>> On Tue, 18 Apr 2023 at 10:40, Joshua C. Colp  wrote:
>>
>>> It's probably best if you read the logic[1]. There's an entire comment
>>> that talks about how it works.
>>>
>>> [1]
>>> https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158
>>>
>>> On Mon, Apr 17, 2023 at 7:10 PM David Cunningham <
>>> dcunning...@voisonics.com> wrote:
>>>
>>>> Hi Joshua,
>>>>
>>>> Could you confirm if the 5 second period for learning a new audio
>>>> stream is a minimum or a maximum? The unusual call flow in question results
>>>> in Asterisk learning a new audio stream when we don't want it to, and
>>>> having a minimum of say 2 seconds of audio would help avoid this.
>>>>
>>>> Thank you!
>>>>
>>>>
>>>> On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp  wrote:
>>>>
>>>>> On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp 
>>>>> wrote:
>>>>>
>>>>>> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham <
>>>>>> dcunning...@voisonics.com> wrote:
>>>>>>
>>>>>>> Hello,
>>>>>>>
>>>>>>> Does anyone know if one of the "strictrtp" options disables RTP
>>>>>>> learning? As far as I can tell from the documentation the values "no" 
>>>>>>> and
>>>>>>> "seqno" are more permissive in allowing other sources rather than less, 
>>>>>>> but
>>>>>>> I thought I'd check.
>>>>>>>
>>>>>>
>>>>>> Setting it to "no" disables the learning.
>>>>>>
>>>>>
>>>>> Since I haven't gotten the email yet I'll just reply to my own.
>>>>>
>>>>> The "no" option disables strict RTP protection. Learning is part of
>>>>> strict RTP protection, it is what determines what the source of media is
>>>>> and then blocks other packets. There is no ability to set it
>>>>> per-peer/per-endpoint.
>>>>>
>>>>> --
>>>>> Joshua C. Colp
>>>>> Asteri

Re: [asterisk-users] RTP address learning and timing problem

[David Cunningham]

Hi Joshua,

Thank you for that. From the code it kind of looks like
STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum:

if (!ast_sockaddr_isnull(>strict_rtp_address)
&& STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(),
rtp->rtp_source_learn.start)) {
ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address
%s\n",

Our call shows:

# grep C-00024cd5 full.log | egrep 'Strict RTP'
[Feb 22 11:16:41] VERBOSE[29023][C-00024cd5] res_rtp_asterisk.c:>
0x2b308c074f80 -- Strict RTP learning after remote address set to:
xx.xx.154.111:18578
[Feb 22 11:17:00] VERBOSE[29023][C-00024cd5] res_rtp_asterisk.c:>
0x2b315c01cbc0 -- Strict RTP learning after remote address set to:
xx.xx.0.12:16498
[Feb 22 11:17:00] VERBOSE[28191][C-00024cd5] res_rtp_asterisk.c:>
0x2b308c074f80 -- Strict RTP switching to RTP remote address
xx.xx.154.111:18578 as source
[Feb 22 11:17:00] VERBOSE[28191][C-00024cd5] res_rtp_asterisk.c:>
0x2b308c074f80 -- Strict RTP learning complete - Locking on source address
xx.xx.154.111:18578
[Feb 22 11:17:00] VERBOSE[28194][C-00024cd5] res_rtp_asterisk.c:>
0x2b315c01cbc0 -- Strict RTP switching source address to xx.xx.114.237:16498
[Feb 22 11:17:01] VERBOSE[28194][C-00024cd5] res_rtp_asterisk.c:>
0x2b315c01cbc0 -- Strict RTP learning complete - Locking on source address
xx.xx.114.237:16498

I'm a bit confused because the second "Strict RTP learning after remote
address set" should reset the rtp_source_learn.start timestamp, and yet the
"Strict RTP learning complete" messages are less than 5000ms after that.
What could be happening?

Thanks again.


On Tue, 18 Apr 2023 at 10:40, Joshua C. Colp  wrote:

> It's probably best if you read the logic[1]. There's an entire comment
> that talks about how it works.
>
> [1]
> https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158
>
> On Mon, Apr 17, 2023 at 7:10 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Joshua,
>>
>> Could you confirm if the 5 second period for learning a new audio stream
>> is a minimum or a maximum? The unusual call flow in question results in
>> Asterisk learning a new audio stream when we don't want it to, and having a
>> minimum of say 2 seconds of audio would help avoid this.
>>
>> Thank you!
>>
>>
>> On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp  wrote:
>>
>>> On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp 
>>> wrote:
>>>
>>>> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham <
>>>> dcunning...@voisonics.com> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> Does anyone know if one of the "strictrtp" options disables RTP
>>>>> learning? As far as I can tell from the documentation the values "no" and
>>>>> "seqno" are more permissive in allowing other sources rather than less, 
>>>>> but
>>>>> I thought I'd check.
>>>>>
>>>>
>>>> Setting it to "no" disables the learning.
>>>>
>>>
>>> Since I haven't gotten the email yet I'll just reply to my own.
>>>
>>> The "no" option disables strict RTP protection. Learning is part of
>>> strict RTP protection, it is what determines what the source of media is
>>> and then blocks other packets. There is no ability to set it
>>> per-peer/per-endpoint.
>>>
>>> --
>>> Joshua C. Colp
>>> Asterisk Project Lead
>>> Sangoma Technologies
>>> Check us out at www.sangoma.com and www.asterisk.org
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>&g

Re: [asterisk-users] RTP address learning and timing problem

[David Cunningham]

Hi Joshua,

Could you confirm if the 5 second period for learning a new audio stream is
a minimum or a maximum? The unusual call flow in question results in
Asterisk learning a new audio stream when we don't want it to, and having a
minimum of say 2 seconds of audio would help avoid this.

Thank you!


On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp  wrote:

> On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp  wrote:
>
>> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham <
>> dcunning...@voisonics.com> wrote:
>>
>>> Hello,
>>>
>>> Does anyone know if one of the "strictrtp" options disables RTP
>>> learning? As far as I can tell from the documentation the values "no" and
>>> "seqno" are more permissive in allowing other sources rather than less, but
>>> I thought I'd check.
>>>
>>
>> Setting it to "no" disables the learning.
>>
>
> Since I haven't gotten the email yet I'll just reply to my own.
>
> The "no" option disables strict RTP protection. Learning is part of strict
> RTP protection, it is what determines what the source of media is and then
> blocks other packets. There is no ability to set it per-peer/per-endpoint.
>
> --
> Joshua C. Colp
> Asterisk Project Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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Re: [asterisk-users] RTP address learning and timing problem

[David Cunningham]

Thank you Joshua.


On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp  wrote:

> On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp  wrote:
>
>> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham <
>> dcunning...@voisonics.com> wrote:
>>
>>> Hello,
>>>
>>> Does anyone know if one of the "strictrtp" options disables RTP
>>> learning? As far as I can tell from the documentation the values "no" and
>>> "seqno" are more permissive in allowing other sources rather than less, but
>>> I thought I'd check.
>>>
>>
>> Setting it to "no" disables the learning.
>>
>
> Since I haven't gotten the email yet I'll just reply to my own.
>
> The "no" option disables strict RTP protection. Learning is part of strict
> RTP protection, it is what determines what the source of media is and then
> blocks other packets. There is no ability to set it per-peer/per-endpoint.
>
> --
> Joshua C. Colp
> Asterisk Project Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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Re: [asterisk-users] RTP address learning and timing problem

[David Cunningham]

Hi Joshua,

Thanks for that. The naming is a little confusing as "no'' makes it sound
like it's "not strict" - good to know though. Is it possible to set
strictrtp to no for just one peer?


On Wed, 1 Mar 2023 at 02:57, Joshua C. Colp  wrote:

> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>> Does anyone know if one of the "strictrtp" options disables RTP learning?
>> As far as I can tell from the documentation the values "no" and "seqno" are
>> more permissive in allowing other sources rather than less, but I thought
>> I'd check.
>>
>
> Setting it to "no" disables the learning.
>
> --
> Joshua C. Colp
> Asterisk Project Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RTP address learning and timing problem

[David Cunningham]

Hello,

Does anyone know if one of the "strictrtp" options disables RTP learning?
As far as I can tell from the documentation the values "no" and "seqno" are
more permissive in allowing other sources rather than less, but I thought
I'd check.

Thanks.


On Thu, 23 Feb 2023 at 12:13, David Cunningham 
wrote:

> Hello,
>
> We have a system that interoperates with an external service, so that the
> basic call flow is:
>
> PSTN origination -> Asterisk A -> External service -> Asterisk B
>
> Initially the SDP from the external service tells the two Asterisks to
> send RTP directly to each other. Part way through the call the external
> service sends re-INVITEs both Asterisks to change the address for audio to
> itself, but this fails to work intermittently. The problem seems to be one
> of timing.
>
> If there's no RTP between the two re-INVITEs then it works fine, and both
> Asterisks send future RTP to the external service as instructed.
>
> The problem is if RTP is transmitted/received in the fraction of the
> second between the two re-INVITEs. If Asterisk A receives the re-INVITE
> first, and then receives RTP from Asterisk B (which hasn't yet received its
> re-INVITE), then it re-learns the media address of Asterisk B and sends
> audio there instead of the new address. Asterisk B gets the second
> re-INVITE with the new media address, but soon re-learns the media address
> of Asterisk A because it's getting RTP from it.
>
> Note we have "canreinvite = no" in sip.conf, but I don't think that's
> relevant to the problem.
>
> Can anyone suggest how to prevent this problem? Is it possible to turn off
> learning the media address per call or per peer?
>
> Thanks for your help.
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
>


-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] RTP address learning and timing problem

[David Cunningham]

Hello,

We have a system that interoperates with an external service, so that the
basic call flow is:

PSTN origination -> Asterisk A -> External service -> Asterisk B

Initially the SDP from the external service tells the two Asterisks to send
RTP directly to each other. Part way through the call the external service
sends re-INVITEs both Asterisks to change the address for audio to itself,
but this fails to work intermittently. The problem seems to be one of
timing.

If there's no RTP between the two re-INVITEs then it works fine, and both
Asterisks send future RTP to the external service as instructed.

The problem is if RTP is transmitted/received in the fraction of the second
between the two re-INVITEs. If Asterisk A receives the re-INVITE first, and
then receives RTP from Asterisk B (which hasn't yet received its
re-INVITE), then it re-learns the media address of Asterisk B and sends
audio there instead of the new address. Asterisk B gets the second
re-INVITE with the new media address, but soon re-learns the media address
of Asterisk A because it's getting RTP from it.

Note we have "canreinvite = no" in sip.conf, but I don't think that's
relevant to the problem.

Can anyone suggest how to prevent this problem? Is it possible to turn off
learning the media address per call or per peer?

Thanks for your help.

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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Re: [asterisk-users] mailing list working?

[David Rebarchik]

Yes, the technical part of the list is working, but I'm not as sure 
about the functional part. (Meaning that several people's questions are 
going unanswered.  I wish that I had the answers they are looking for, 
but alas, I don't.)


Dave

On 1/16/2023 6:08 AM, marek wrote:

there are new versions of Asterisk but mailing list is empty

http://lists.digium.com/pipermail/asterisk-users/

Marek





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Re: [asterisk-users] Voicemail Transcription with openai/whisper

[David Rebarchik]

I really love this idea. Thanks for sharing. I've been looking for a 
good way to provide this service to my customers. Hopefully this will 
work for me too.


Thanks,
Dave

On 11/27/2022 8:08 AM, Doug Lytle wrote:

Everybody,

I've recently discovered openai/whisper and have been trying in 
earnest to get this working with Asterisk for voicemail transcriptions 
(Currently using the NerdVittles script with IBM Watson)


https://github.com/openai/whisper

After spending several hours today, I've successfully integrated my 
home Asterisk 16 voicemail with Whisper.


Once I have followed the instructions for setting up an API server

https://blog.deepgram.com/how-to-build-an-openai-whisper-api/

Initially, I setup a quad core VM to test this with, but discovered 
that without a dedicated card for the inference that it was horribly 
slow.  So, I've set up testing on my desktop (Kubuntu 20) since I have 
an nVidia GTX 1060 installed.


For the integration with Asterisk, I'm using a slightly modified 
script from nerdvittles IBM Watson script


sendmailibm

That can be found on their website

https://nerdvittles.com/free-asterisk-voicemail-transcription-with-ibms-stt-engine/

I will probably find a low cost nVidia video card and get a stand 
alone Linux box running to handle this project.


If you're interested in the details, let me know.

Doug




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Re: [asterisk-users] CSeq reset on re-INVITE

[David Cunningham]

Okay, thanks very much for your help Joshua.


On Mon, 31 Oct 2022 at 10:07, Joshua C. Colp  wrote:

> On Sun, Oct 30, 2022 at 5:00 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Joshua,
>>
>> Thanks very much. I presume this is the relevant part:
>> "strictly monotonically increasing and contiguous CSeq sequence numbers
>> (increasing-by-one) in each direction"
>>
>> In that case I wonder what could be causing the 404 Not Found error. I've
>> attached the relevant SIP packets from the Asterisk log. Can anyone see an
>> issue that would cause the error?
>>
>
> Based on the provided trace the signaling appears correct. I don't think
> there's anything on the Asterisk side wrong, and don't think it'll lead to
> an answer.
>
> --
> Joshua C. Colp
> Asterisk Project Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
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Re: [asterisk-users] CSeq reset on re-INVITE

[David Cunningham]

Hi Joshua,

Thanks very much. I presume this is the relevant part:
"strictly monotonically increasing and contiguous CSeq sequence numbers
(increasing-by-one) in each direction"

In that case I wonder what could be causing the 404 Not Found error. I've
attached the relevant SIP packets from the Asterisk log. Can anyone see an
issue that would cause the error?

Thanks in advance.


On Sat, 29 Oct 2022 at 12:03, Joshua C. Colp  wrote:

> On Fri, Oct 28, 2022 at 6:28 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>> We have a problem where Asterisk is resetting the CSeq on a re-INVITE,
>> and the phone receiving the re-INVITE is rejecting it, probably as a result
>> of that. Would anyone be able to offer any insight please?
>>
>> The scenario is:
>>
>> Phone A makes call 1 to Asterisk which dials call 2 to phone B, which
>> answers the call.
>>
>> Phone B puts call 1 on hold, makes call 3 to Asterisk which dials call 4
>> to phone C, which answers the call.
>>
>> Phone B does an attended REFER transfer of call 2 to call 3, taking
>> itself out of the call. Asterisk bridges the remaining calls, so phones A
>> and C are now talking to each other.
>>
>> Asterisk sends a re-INVITE to phone A with a P-Asserted-Identity, to tell
>> phone A the updated details of phone C that it's talking to. However phone
>> A rejects the re-INVITE with a "404 Not found" error.
>>
>> The only explanation I can see for the "404 Not found" error is that call
>> 1 was set up with "CSeq: 954698786 INVITE", whereas the re-INVITE Asterisk
>> sends with the P-Asserted-Identity has "CSeq: 102 INVITE". Why is Asterisk
>> resetting the CSeq on the re-INVITE, and doesn't this appear to be
>> incorrect?
>>
>
> It's not incorrect. Each direction has its own CSeq[1]. From Phone A to
> Asterisk can be 954698786 and from Asterisk to Phone A can be 102.
>
> [1] https://www.rfc-editor.org/rfc/rfc3261#section-12.2.1.1
>
> --
> Joshua C. Colp
> Asterisk Project Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782

ORIGINAL CALL PHONE A TO ASTERISK:

[Oct 27 16:17:43] VERBOSE[2051] chan_sip.c: 
<--- SIP read from UDP:111.111.52.208:5060 --->
INVITE sip:22@111.111.52.201:5060;transport=udp SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 
111.111.52.208;branch=z9hG4bK2d9a.a2b48027e41283fbb0d79a59edd6d1ef.0
Via: SIP/2.0/UDP 222.222.127.193:5060;rport=5060;branch=z9hG4bK4ee82f350717c20f6
Proxy-Authorization: Digest 
username="11",realm="example.com",nonce="Y1oVo2NaFHcQ7EwgMuEk3/LGyhpkbC8t",uri="sip:222...@example.com:5060;user=phone",response="636c2526a687a35d2683951d0be695d4"
Max-Forwards: 69
From: "11" ;tag=8048438a0b
To: 
Call-ID: 99eab43e2d6c6a20
CSeq: 954698786 INVITE


ASTERISK SENDS CALL TO PHONE B:

[Oct 27 16:17:43] VERBOSE[1205][C-002558c4] chan_sip.c: Reliably Transmitting 
(NAT) to 111.111.52.208:5060:
INVITE sip:22@111.111.52.208:5060 SIP/2.0
Via: SIP/2.0/UDP 111.111.52.201:5060;branch=z9hG4bK39357a26;rport
Max-Forwards: 70
From: "11" ;tag=as3569276f
To: 
Contact: 
Call-ID: 407ff6cd243025df68a25415741bb5a0@111.111.52.201:5060


PHONE B CREATES NEW CALL TO PHONE C:

[Oct 27 16:17:51] VERBOSE[2051] chan_sip.c: 
<--- SIP read from UDP:111.111.52.208:5060 --->
INVITE sip:33@111.111.52.201:5060;transport=udp SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 
111.111.52.208;branch=z9hG4bKa1b8.04fe0a3c5e61a15fa889b1d3c087e21d.0
Via: SIP/2.0/UDP 
10.10.10.112;rport=60608;received=333.333.96.250;branch=z9hG4bK9bfc44c3c33f64f74
Proxy-Authorization: Digest 
username="22",realm="example.com",nonce="Y1oVq2NaFH+v2ge8RzGSe6aI+Y1BIOQ/",uri="sip:333...@example.com:5060;user=phone",response="571b389bf09d3bc6fb3d042720435923"
Max-Forwards: 69
From: "User User" ;tag=ec8ebeab1f
To: 
Call-ID: 84fb5d56a369394a
CSeq: 1053485297 INVITE


PHONE B REFERS FIRST CALL TO SECOND:

[Oct 27 16:17:58] VERBOSE[2051] chan_sip.c: 
<--- SIP read from UDP:111.111.52.208:5060 --->
REFER sip:111

[asterisk-users] CSeq reset on re-INVITE

[David Cunningham]

Hello,

We have a problem where Asterisk is resetting the CSeq on a re-INVITE, and
the phone receiving the re-INVITE is rejecting it, probably as a result of
that. Would anyone be able to offer any insight please?

The scenario is:

Phone A makes call 1 to Asterisk which dials call 2 to phone B, which
answers the call.

Phone B puts call 1 on hold, makes call 3 to Asterisk which dials call 4 to
phone C, which answers the call.

Phone B does an attended REFER transfer of call 2 to call 3, taking itself
out of the call. Asterisk bridges the remaining calls, so phones A and C
are now talking to each other.

Asterisk sends a re-INVITE to phone A with a P-Asserted-Identity, to tell
phone A the updated details of phone C that it's talking to. However phone
A rejects the re-INVITE with a "404 Not found" error.

The only explanation I can see for the "404 Not found" error is that call 1
was set up with "CSeq: 954698786 INVITE", whereas the re-INVITE Asterisk
sends with the P-Asserted-Identity has "CSeq: 102 INVITE". Why is Asterisk
resetting the CSeq on the re-INVITE, and doesn't this appear to be
incorrect?

Thanks in advance for any help.

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
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[asterisk-users] VoIP support engineer opportunity

[David Cunningham]

Hello,

Voisonics is hiring a VoIP support engineer to assist our customers running
Asterisk based hosted PBX platforms. This is a part-time contract
work-from-home position.

For communication reasons we're looking for someone in a timezone
encompassing New Zealand, Canada, the USA, and Mexico. If you are not
physically located in that area please do not apply - being "flexible" from
another part of the world is not what we're looking for.

The role involves providing technical support of Asterisk based PBX
platforms to our customer's technical staff, Linux system administration,
and small dev-ops type development projects. It does not involve providing
technical support to end users or the general public.

Customers are located around the world. You will generally be responding
during your business hours, though sometimes out of hours work will be
necessary. Once training is completed, the position will involve providing
24x7 on-call emergency cover in rotation with other staff.

Must-have:
1. Fluent command-line Linux ability on Ubuntu, Debian, Rocky, CentOS,
and/or RedHat.
2. Asterisk administration and configuration experience.
3. SIP debugging experience. For example, you should know what packets are
typically involved in setting up a call.
4. Experience with administration and configuration of Apache and MySQL or
PostgreSQL.
5. Ability to program in Perl or shell scripts.
6. Good written and verbal English language ability.
7. Ability to solve problems and create solutions independently. Although
there are other staff, most work is done by one engineer on their own.
8. Have experience providing professional IT support to business.
9. Be an individual self-employed contractor.

Nice-to-have:
1. Experience with Kamailio, NFS, GlusterFS, Puppet, or Zabbix.
2. C, Go, Javascript, AngularJS, HTML or CSS programming ability.
3. Advanced network knowledge (beyond basic Linux networking which is a
must-have).

To apply for this role please email me off-list. Do not call by phone
unless arranged in advance.

In your application:
1. List your experience/compatibility for each of the must-have
requirements individually, plus any of the nice-to-have items you fit as
well.
2. Provide your physical location, hours of availability, and indication of
hourly rate.
3. Let us know what other work you already have during your hours of
availability.
4. A full CV is welcome.

Thank you,

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
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Re: [asterisk-users] Asterisk "we couldn't allocate a port for RTP" errors

[David Cunningham]

Hi Joel,

We see some channels stuck on the StopMonitor application which could be
the cause, though there are only about 20 of them.

An Asterisk restart is the immediate cure for sure - we were hoping to
identify some cause in order to prevent a repeat.

Thanks very much for your advice.


On Thu, 28 Jul 2022 at 13:31, Joel Serrano  wrote:

> I would check if you don't have any channels in a hung/zombie state...
>
> Have a look if "core show calls" matches "core show channels".
>
> Either way, it seems wonky, so you might end up having to give that
> asterisk a restart... :S
>
> On Wed, Jul 27, 2022 at 6:21 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>> We have an Asterisk 13.38.2 server which today started giving "we
>> couldn't allocate a port for RTP" errors. The output of "netstat -anp"
>> showed that Asterisk was using all 10,000 ports allocated for RTP, even
>> though it only had a maximum of around 200 concurrent calls at any point in
>> the day. Even now when calls have fallen to less than 30, there are still
>> almost 10,000 ports in use.
>>
>> Does anyone have any suggestions on how to find out why so many ports are
>> in use?
>>
>> I'm aware that Asterisk 13 is no longer supported by Sangoma so I haven't
>> opened a bug report.
>>
>> Thanks in advance for any advice.
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk "we couldn't allocate a port for RTP" errors

[David Cunningham]

Hello,

We have an Asterisk 13.38.2 server which today started giving "we couldn't
allocate a port for RTP" errors. The output of "netstat -anp" showed that
Asterisk was using all 10,000 ports allocated for RTP, even though it only
had a maximum of around 200 concurrent calls at any point in the day. Even
now when calls have fallen to less than 30, there are still almost 10,000
ports in use.

Does anyone have any suggestions on how to find out why so many ports are
in use?

I'm aware that Asterisk 13 is no longer supported by Sangoma so I haven't
opened a bug report.

Thanks in advance for any advice.

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] TCP dial via proxy

[David Cunningham]

Hi Łukasz,

A TCP call works fine under normal circumstances. It's just when we send
the call via a proxy that we have a problem, because the call to the proxy
doesn't appear to use TCP.

Thank you.


On Fri, 22 Jul 2022 at 11:58, Łukasz Grzywański 
wrote:

> Hi,
> which version are you using ?
> please show: asterisk -rx "sip show peer sip-peer"
>
> I checked...
> I use UDP and TCP, my phone via UDP, telekom via TCP and works
>
>
> same  => n,dial(SIP/${EXTEN}@sip-trunk-telekom)
>
> [image: image.png]
>
>
> On Thu, 21 Jul 2022 at 23:58, David Cunningham 
> wrote:
>
>> Thank you Thomas. I know it would be good to move to pjsip, and that's
>> coming in a future product version, but it isn't used in the version of
>> this scenario.
>>
>>
>> On Fri, 22 Jul 2022 at 01:30, Thomas Ray 
>> wrote:
>>
>>> The answer is chan_pjsip. You can do this with chan_pjsip. There’s no
>>> real support for chan_sip anymore. It’s dead, it’s going away. No fixes or
>>> updates will be accepted against it as of this point.
>>>
>>>
>>>
>>> *From: *asterisk-users  on
>>> behalf of Dovid Bender 
>>> *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion <
>>> asterisk-users@lists.digium.com>
>>> *Date: *Thursday, July 21, 2022 at 9:21 AM
>>> *To: *Asterisk Users Mailing List - Non-Commercial Discussion <
>>> asterisk-users@lists.digium.com>
>>> *Subject: *Re: [asterisk-users] TCP dial via proxy
>>>
>>>
>>>
>>> David,
>>>
>>>
>>>
>>> We had this exact "issue" in the past and were not able to figure out
>>> how to do it. Where we wanted tcp we prefixed the sip URI with "force_tcp".
>>> So:
>>>
>>> Dial(SIP/1234@1.1.1.1//2.2.2.2 <http://1234@1.1.1.1/2.2.2.2>)
>>>
>>> became:
>>>
>>> Dial(SIP/force_tcp1234@1.1.1.1//2.2.2.2
>>> <http://force_tcp1234@1.1.1.1/2.2.2.2>)
>>>
>>> On Kamailio's side in the FORWARD block we added:
>>>
>>> # HACK for forcing TCP
>>> if ($oU != $null && $(oU{s.len}) != 0) {
>>> $var(prefix) = $(oU{s.substr,0,9});
>>> if ($var(prefix) == "force_tcp") {
>>> $rU = $(oU{s.substr,9,0});
>>> add_uri_param( "transport=tcp" );
>>> $fs = "tcp:" + $Ri + ":5060";
>>> }
>>> }
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Jul 20, 2022 at 10:47 PM David Cunningham <
>>> dcunning...@voisonics.com> wrote:
>>>
>>> Hello,
>>>
>>>
>>>
>>> We have an Asterisk dial which sends the call via a proxy using //, for
>>> example:
>>>
>>>
>>>
>>> Dial(SIP/${EXTEN}@peer_address//proxy_address)
>>>
>>>
>>>
>>> Does anyone know how we can make the SIP to the proxy use TCP? We tried
>>> making proxy_address match a peer in sip.conf with "transport = tcp" but
>>> that didn't seem to work. We are using chan_sip.
>>>
>>>
>>>
>>> Thanks very much for any advice.
>>>
>>>
>>>
>>> --
>>>
>>> David Cunningham, Voisonics Limited
>>> http://voisonics.com/
>>> USA: +1 213 221 1092
>>> New Zealand: +64 (0)28 2558 3782
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>> -- _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/ New to Asterisk? Start here:
>>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>> asterisk-users mailing

Re: [asterisk-users] TCP dial via proxy

[David Cunningham]

Thank you Thomas. I know it would be good to move to pjsip, and that's
coming in a future product version, but it isn't used in the version of
this scenario.


On Fri, 22 Jul 2022 at 01:30, Thomas Ray  wrote:

> The answer is chan_pjsip. You can do this with chan_pjsip. There’s no real
> support for chan_sip anymore. It’s dead, it’s going away. No fixes or
> updates will be accepted against it as of this point.
>
>
>
> *From: *asterisk-users  on
> behalf of Dovid Bender 
> *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Date: *Thursday, July 21, 2022 at 9:21 AM
> *To: *Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject: *Re: [asterisk-users] TCP dial via proxy
>
>
>
> David,
>
>
>
> We had this exact "issue" in the past and were not able to figure out how
> to do it. Where we wanted tcp we prefixed the sip URI with "force_tcp". So:
>
> Dial(SIP/1234@1.1.1.1//2.2.2.2 <http://1234@1.1.1.1/2.2.2.2>)
>
> became:
>
> Dial(SIP/force_tcp1234@1.1.1.1//2.2.2.2
> <http://force_tcp1234@1.1.1.1/2.2.2.2>)
>
> On Kamailio's side in the FORWARD block we added:
>
> # HACK for forcing TCP
> if ($oU != $null && $(oU{s.len}) != 0) {
> $var(prefix) = $(oU{s.substr,0,9});
> if ($var(prefix) == "force_tcp") {
> $rU = $(oU{s.substr,9,0});
> add_uri_param( "transport=tcp" );
> $fs = "tcp:" + $Ri + ":5060";
> }
> }
>
>
>
>
>
>
>
> On Wed, Jul 20, 2022 at 10:47 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
> Hello,
>
>
>
> We have an Asterisk dial which sends the call via a proxy using //, for
> example:
>
>
>
> Dial(SIP/${EXTEN}@peer_address//proxy_address)
>
>
>
> Does anyone know how we can make the SIP to the proxy use TCP? We tried
> making proxy_address match a peer in sip.conf with "transport = tcp" but
> that didn't seem to work. We are using chan_sip.
>
>
>
> Thanks very much for any advice.
>
>
>
> --
>
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> -- _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/ New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users
> mailing list To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> _____
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] TCP dial via proxy

[David Cunningham]

Hi Dovid,

Thanks for the reply. We are indeed able to force TCP from the Kamailio
proxy, but haven't been able to force it between Asterisk and Kamailio.


On Fri, 22 Jul 2022 at 01:21, Dovid Bender  wrote:

> David,
>
> We had this exact "issue" in the past and were not able to figure out how
> to do it. Where we wanted tcp we prefixed the sip URI with "force_tcp". So:
> Dial(SIP/1234@1.1.1.1//2.2.2.2)
> became:
> Dial(SIP/force_tcp1234@1.1.1.1//2.2.2.2)
> On Kamailio's side in the FORWARD block we added:
> # HACK for forcing TCP
> if ($oU != $null && $(oU{s.len}) != 0) {
> $var(prefix) = $(oU{s.substr,0,9});
> if ($var(prefix) == "force_tcp") {
> $rU = $(oU{s.substr,9,0});
> add_uri_param( "transport=tcp" );
> $fs = "tcp:" + $Ri + ":5060";
> }
> }
>
>
>
> On Wed, Jul 20, 2022 at 10:47 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>> We have an Asterisk dial which sends the call via a proxy using //, for
>> example:
>>
>> Dial(SIP/${EXTEN}@peer_address//proxy_address)
>>
>> Does anyone know how we can make the SIP to the proxy use TCP? We tried
>> making proxy_address match a peer in sip.conf with "transport = tcp" but
>> that didn't seem to work. We are using chan_sip.
>>
>> Thanks very much for any advice.
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



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Re: [asterisk-users] TCP dial via proxy

[David Cunningham]

Hi Henning,

We tried using outboundproxy as follows, but the SIP from Asterisk to the
proxy still went via UDP. Is there anything else you'd suggest? Thank you.

In extensions.conf:

Dial(SIP/${EXTEN}@sip-peer)

In sip.conf:

[general]
tcpenable = yes
tcpbindaddr = 0.0.0.0

[sip-peer]
host = final.destination.com
transport = tcp
outboundproxy = our.proxy.com


On Fri, 22 Jul 2022 at 01:23, Henning Follmann 
wrote:

> On Thu, Jul 21, 2022 at 02:46:07PM +1200, David Cunningham wrote:
> > Hello,
> >
> > We have an Asterisk dial which sends the call via a proxy using //, for
> > example:
> >
> > Dial(SIP/${EXTEN}@peer_address//proxy_address)
> >
> > Does anyone know how we can make the SIP to the proxy use TCP? We tried
> > making proxy_address match a peer in sip.conf with "transport = tcp" but
> > that didn't seem to work. We are using chan_sip.
> >
> > Thanks very much for any advice.
> >
>
> Have you tried to define
> outboundproxy=proxy_address
> in your sip.conf?
>
> -H
>
>
>
> --
> Henning Follmann   | hfollm...@itcfollmann.com
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



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[asterisk-users] TCP dial via proxy

[David Cunningham]

Hello,

We have an Asterisk dial which sends the call via a proxy using //, for
example:

Dial(SIP/${EXTEN}@peer_address//proxy_address)

Does anyone know how we can make the SIP to the proxy use TCP? We tried
making proxy_address match a peer in sip.conf with "transport = tcp" but
that didn't seem to work. We are using chan_sip.

Thanks very much for any advice.

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
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Re: [asterisk-users] Listen on 2 of 3 IP addresses

[David Cunningham]

Thank you Thomas.


On Mon, 18 Jul 2022 at 12:24, Thomas Ray  wrote:

> Moving to chan_pjsip solves this problem.
>
>
>
> *From: *asterisk-users  on
> behalf of David Cunningham 
> *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Date: *Sunday, July 17, 2022 at 5:53 PM
> *To: *Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject: *Re: [asterisk-users] Listen on 2 of 3 IP addresses
>
>
>
> Thank you for that reply, Joshua.
>
>
>
> Henning, we can restrict access using the firewall, but unfortunately that
> doesn't solve the problem with the address that's put in the SDP.
>
>
>
>
>
> On Sat, 16 Jul 2022 at 03:14, Henning Follmann 
> wrote:
>
> On Fri, Jul 15, 2022 at 08:57:46AM -0300, Joshua C. Colp wrote:
> > On Fri, Jul 15, 2022 at 1:37 AM David Cunningham <
> dcunning...@voisonics.com>
> > wrote:
> >
> > > Hello,
> > >
> > > We have an Asterisk server with 3 IP addresses, and need to listen on
> only
> > > 2 of those. This is with chan_sip. Does anyone know if it's possible?
> > >
> > > If Asterisk listens on the third address then it seems to cause
> problems
> > > with the media address put in the SDP for our use case.
> > >
> >
> > It's not. The chan_sip module allows you to bind to one thing, either a
> > specific address or an any address.
> >
>
> Well...
> maybe chan_sip cannot, but your OS can restrict traffic on any port/iface.
>
> -H
>
> --
> Henning Follmann   | hfollm...@itcfollmann.com
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
>
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
>
> -- _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/ New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users
> mailing list To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
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http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
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Re: [asterisk-users] Listen on 2 of 3 IP addresses

[David Cunningham]

Thank you for that reply, Joshua.

Henning, we can restrict access using the firewall, but unfortunately that
doesn't solve the problem with the address that's put in the SDP.


On Sat, 16 Jul 2022 at 03:14, Henning Follmann 
wrote:

> On Fri, Jul 15, 2022 at 08:57:46AM -0300, Joshua C. Colp wrote:
> > On Fri, Jul 15, 2022 at 1:37 AM David Cunningham <
> dcunning...@voisonics.com>
> > wrote:
> >
> > > Hello,
> > >
> > > We have an Asterisk server with 3 IP addresses, and need to listen on
> only
> > > 2 of those. This is with chan_sip. Does anyone know if it's possible?
> > >
> > > If Asterisk listens on the third address then it seems to cause
> problems
> > > with the media address put in the SDP for our use case.
> > >
> >
> > It's not. The chan_sip module allows you to bind to one thing, either a
> > specific address or an any address.
> >
>
> Well...
> maybe chan_sip cannot, but your OS can restrict traffic on any port/iface.
>
> -H
>
> --
> Henning Follmann   | hfollm...@itcfollmann.com
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
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http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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_
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[asterisk-users] Listen on 2 of 3 IP addresses

[David Cunningham]

Hello,

We have an Asterisk server with 3 IP addresses, and need to listen on only
2 of those. This is with chan_sip. Does anyone know if it's possible?

If Asterisk listens on the third address then it seems to cause problems
with the media address put in the SDP for our use case.

Thanks very much,

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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Re: [asterisk-users] No audio for 10 seconds and then comfort noise

[David Cunningham]

Hi Joshua,

You're right, it was a firewall problem. One of those things where testing
a change in one place throws up a previously unseen problem somewhere else!
Thanks for the tip.


On Thu, 19 May 2022 at 21:18, Joshua C. Colp  wrote:

> On Thu, May 19, 2022 at 6:04 AM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Dovid and Joshua,
>>
>> The PSTN is sending RTP immediately after the 200 OK, on both legs of the
>> call. Since the PCAP taken on the Asterisk server itself shows this RTP
>> from the PSTN then presumably it can't be a network issue preventing the
>> RTP.
>>
>> Having said that, the problem is not reproduced when the peer is another
>> Asterisk server on the same network, and that does point to a network
>> difference.
>>
>> Is there any other way in which the RTP keepalive might affect Asterisk's
>> behaviour?
>>
>
> No, the option only does anything if no RTP has been sent for a period of
> time. It doesn't fundamentally alter the behavior of RTP in general.
>
> Another thing to consider is that a PCAP is taken before any local
> firewall rules are applied, which can give a false impression that the
> firewall on the system is not an issue when in reality it can be. That's
> something to check.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
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http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
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Re: [asterisk-users] No audio for 10 seconds and then comfort noise

[David Cunningham]

Hi Dovid and Joshua,

The PSTN is sending RTP immediately after the 200 OK, on both legs of the
call. Since the PCAP taken on the Asterisk server itself shows this RTP
from the PSTN then presumably it can't be a network issue preventing the
RTP.

Having said that, the problem is not reproduced when the peer is another
Asterisk server on the same network, and that does point to a network
difference.

Is there any other way in which the RTP keepalive might affect Asterisk's
behaviour?

Thanks for your help on this.


On Thu, 19 May 2022 at 20:40, Joshua C. Colp  wrote:

> On Thu, May 19, 2022 at 3:52 AM Dovid Bender  wrote:
>
>> David,
>>
>> Are you getting any RTP from the PSTN for either leg? If not it could be
>> that they assume you are behind NAT and want to see where the SRC of the
>> RTP before they send it back. We had a few carriers that did this. The
>> easiest way to get around it was to play a 0.5 second audio clip to the
>> incoming leg. This will send RTP to the inbound carrier, causing them to
>> send RTP back to you which would then hit the terminating carrier, which
>> then sends you back RTP completing the loop. The dialplan looks
>> something like this.
>>
>> same =>n, Progress()
>> same =>n,
>> Playback(/var/lib//asterisk_custom/sounds/xc,noanswer)
>> same =>n, Dial(SIP/+${EXTEN}@carrier,,)
>>
>
> I've also seen this happen due to networking equipment, specifically the
> equipment wanting Asterisk to send packets before allowing packets in. If
> both sides of the call are in this state, then you reach a stalemate and
> media won't flow. Since rtp_keepalive is generated by Asterisk, it gets
> sent, and media starts flowing.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
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USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
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Re: [asterisk-users] No audio for 10 seconds and then comfort noise

[David Cunningham]

We found that the 10 seconds relates to the "rtpkeepalive =10" in our
sip.conf. If the rtpkeepalive is reduced then the delay reduces as well. If
rtpkeepalive is removed from sip.conf then audio never starts flowing.

Does that help anyone make sense of what's happening?

We have DAHDI running on the server:

# asterisk -rx 'dahdi show version'
DAHDI Version: 3.0.0 Echo Canceller:
# asterisk -rx 'dahdi show status'
Description  Alarms  IRQbpviol CRCFra
Codi Options  LBO


On Thu, 19 May 2022 at 15:51, David Cunningham 
wrote:

> Hello,
>
> We are running an Asterisk 13 server which is having a strange problem,
> where on calls which are received from the PSTN and then forwarded out to
> the PSTN again there is no audio for the first 10 seconds of the call. At
> the 10 second mark audio starts flowing fine, and in a PCAP we see that it
> starts with a few "comfort noise" packers before the real audio starts.
>
> It can be reproduced with a very simple extension like this:
> exten => 4081234567, 2, Dial(SIP/6501234...@bb.bb.bb.138)
> where 4081234567 is the number we receive the call on, and 6501234567 is
> the number we're forwarding it out to.
>
> In the Asterisk log we don't see any obvious reason for the audio to start
> flowing at the 10 second mark. All that is logged at that time is the
> following below.
>
> Would anyone have any ideas? Historically Asterisk didn't generate comfort
> noise - has that changed in version 13?
>
> [May 17 20:26:24] VERBOSE[11933] res_rtp_asterisk.c: Sent Comfort Noise
> RTP packet to aa.aa.aa.76:64280 (type 02, seq 009268, ts 00, len 01)
> [May 17 20:26:24] VERBOSE[17794][C-0027] res_rtp_asterisk.c: Got RTP
> packet from aa.aa.aa.76:64280 (type 00, seq 000662, ts 105920, len 000160)
> [May 17 20:26:24] DEBUG[17725][C-0027] res_rtp_asterisk.c: Ooh, format
> changed from none to ulaw
> [May 17 20:26:24] DEBUG[17725][C-0027] res_rtp_asterisk.c: Starting
> RTCP transmission on RTP instance '0x14f4cc025998'
> [May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Sent RTP
> packet to bb.bb.bb.20:35412 (type 00, seq 020934, ts 105920, len 000160)
> [May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Got RTP
> packet from bb.bb.bb.20:35412 (type 00, seq 029996, ts 102760, len 000160)
>
> Thanks very much,
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
>


-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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[asterisk-users] No audio for 10 seconds and then comfort noise

[David Cunningham]

Hello,

We are running an Asterisk 13 server which is having a strange problem,
where on calls which are received from the PSTN and then forwarded out to
the PSTN again there is no audio for the first 10 seconds of the call. At
the 10 second mark audio starts flowing fine, and in a PCAP we see that it
starts with a few "comfort noise" packers before the real audio starts.

It can be reproduced with a very simple extension like this:
exten => 4081234567, 2, Dial(SIP/6501234...@bb.bb.bb.138)
where 4081234567 is the number we receive the call on, and 6501234567 is
the number we're forwarding it out to.

In the Asterisk log we don't see any obvious reason for the audio to start
flowing at the 10 second mark. All that is logged at that time is the
following below.

Would anyone have any ideas? Historically Asterisk didn't generate comfort
noise - has that changed in version 13?

[May 17 20:26:24] VERBOSE[11933] res_rtp_asterisk.c: Sent Comfort Noise RTP
packet to aa.aa.aa.76:64280 (type 02, seq 009268, ts 00, len 01)
[May 17 20:26:24] VERBOSE[17794][C-0027] res_rtp_asterisk.c: Got RTP
packet from aa.aa.aa.76:64280 (type 00, seq 000662, ts 105920, len 000160)
[May 17 20:26:24] DEBUG[17725][C-0027] res_rtp_asterisk.c: Ooh, format
changed from none to ulaw
[May 17 20:26:24] DEBUG[17725][C-0027] res_rtp_asterisk.c: Starting
RTCP transmission on RTP instance '0x14f4cc025998'
[May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Sent RTP
packet to bb.bb.bb.20:35412 (type 00, seq 020934, ts 105920, len 000160)
[May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Got RTP
packet from bb.bb.bb.20:35412 (type 00, seq 029996, ts 102760, len 000160)

Thanks very much,

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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Re: [asterisk-users] Context for 302 Moved response

[David Cunningham]

Hi Joshua,

Thanks for the reply. In this case we get a special SIP header in the 302,
but I guess we'll need to find another solution to use it.


On Wed, 27 Apr 2022 at 21:27, Joshua C. Colp  wrote:

> On Wed, Apr 27, 2022 at 5:33 AM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Jon,
>>
>> Thank you for the reply. We wanted to read a particular SIP header in the
>> 302 Moved response, but it seems that Asterisk creates a Local channel for
>> the redirected call and the SIP_HEADER() function isn't available, so we
>> can't really do what we wanted at all.
>>
>
> Neither chan_sip or chan_pjsip provide such ability even if you had access
> to the SIP or PJSIP channel. SIP_HEADER() gets headers from an incoming
> INVITE, same for PJSIP_HEADER().
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
David Cunningham, Voisonics Limited
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USA: +1 213 221 1092
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-- 
_
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Re: [asterisk-users] Context for 302 Moved response

[David Cunningham]

Hi Jon,

Thank you for the reply. We wanted to read a particular SIP header in the
302 Moved response, but it seems that Asterisk creates a Local channel for
the redirected call and the SIP_HEADER() function isn't available, so we
can't really do what we wanted at all.

Thanks anyway!


On Wed, 27 Apr 2022 at 18:57, Jon Bonilla (Manwe) 
wrote:

> El Wed, 27 Apr 2022 12:27:03 +1200
> David Cunningham  escribió:
>
> > Hello,
> >
> > Does anyone know of a way to have a call go to a particular context when
> a
> > 302 Moved is received in response to an invite? This is with chan_sip. We
> > tried setting __TRANSFER_CONTEXT but it didn't seem to have any effect.
> > Basically if a remote device returns a 302 Moved we want to send the call
> > somewhere different to all other calls.
> >
> > Thanks very much,
> >
>
>
> You can detect a 302 in the dialplan. Not perfect but does the job.
>
> same => n,GotoIf($[${EXISTS(${FORWARDERNAME})}]?sipcfu)
>
>
> --
> PekePBX, the multitenant PBX solution
> https://pekepbx.com
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users



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[asterisk-users] Context for 302 Moved response

[David Cunningham]

Hello,

Does anyone know of a way to have a call go to a particular context when a
302 Moved is received in response to an invite? This is with chan_sip. We
tried setting __TRANSFER_CONTEXT but it didn't seem to have any effect.
Basically if a remote device returns a 302 Moved we want to send the call
somewhere different to all other calls.

Thanks very much,

-- 
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Re: [asterisk-users] T38 state values

[David Cunningham]

Hi Joshua,

Thank you for that. In the end it seems to have been a firewall blocking
the UDPTL ports.


On Thu, 24 Mar 2022 at 11:15, Joshua C. Colp  wrote:

> On Wed, Mar 23, 2022 at 7:07 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Joshua,
>>
>> Thanks very much for that information. I guess in that case the 'not
>> working' scenario has a T38 state of 'T38 was negotiated and is active',
>> which doesn't help us find the problem much.
>>
>> In this case the T38 is passthrough, and in the not working scenario in a
>> PCAP we see:
>> PSTN -> Asterisk: 't30ind: cng' packet is received
>> Asterisk -> ATA: 't30ind: cng' is not sent
>>
>> Would you have any suggestion on what could prevent the 't30ind: cng'
>> being passed through? The Asterisk T38 settings in sip.conf are:
>>
>> t38pt_udptl = yes
>> t38_udptl = yes
>> t38pt_rtp = no
>> t38pt_tcp = no
>>
>> Thanks again.
>>
>
> Nope. You could see if increasing the core debug and enabling udptl debug
> on the console sheds any light, otherwise it's just digging into things.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] T38 state values

[David Cunningham]

Hi Joshua,

Thanks very much for that information. I guess in that case the 'not
working' scenario has a T38 state of 'T38 was negotiated and is active',
which doesn't help us find the problem much.

In this case the T38 is passthrough, and in the not working scenario in a
PCAP we see:
PSTN -> Asterisk: 't30ind: cng' packet is received
Asterisk -> ATA: 't30ind: cng' is not sent

Would you have any suggestion on what could prevent the 't30ind: cng' being
passed through? The Asterisk T38 settings in sip.conf are:

t38pt_udptl = yes
t38_udptl = yes
t38pt_rtp = no
t38pt_tcp = no

Thanks again.


On Fri, 18 Mar 2022 at 21:59, Joshua C. Colp  wrote:

> On Fri, Mar 18, 2022 at 12:27 AM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>> We have a problem where one fax ATA connected to Asterisk works, and
>> another ATA with the same model and firmware does not. Both are configured
>> to use T38.
>>
>> Basically the call comes in to Asterisk which then does a Dial to the ATA
>> that's registered via SIP, and once that's set up then there's a re-INVITE
>> from the ATA to switch to T38. The ATA sends a "t30ind: no-signal" after
>> the 200 OK, and both work up until this point. But then in the working
>> scenario Asterisk sends a "t30ind: cng" back to the ATA, and in the
>> not-working scenario Asterisk does not send any T38 data at all.
>>
>> We've noticed that in the working scenario when the 200 OK is sent from
>> Asterisk to the ATA, Asterisk logs:
>>
>> [Mar 17 16:07:50] DEBUG[53838][C-0013e292] chan_sip.c: T38 state changed
>> to 2 on channel SIP/xx.xx.246.70:5060-0030dde1
>> [Mar 17 16:07:50] DEBUG[6862][C-0013e292] chan_sip.c: T38 state changed
>> to 1 on channel SIP/product-local-bw70-0030ddd9
>>
>> Whereas in the not working scenario it logs:
>>
>> [Mar 17 16:23:05] DEBUG[53838][C-0013e39a] chan_sip.c: T38 state changed
>> to 3 on channel SIP/product-local-bw70-0030e0a0
>> [Mar 17 16:23:05] DEBUG[24973][C-0013e39a] chan_sip.c: T38 state changed
>> to 3 on channel SIP/xx.xx.246.70:5060-0030e0a1
>>
>> Notice the difference in the "T38 state changed to" values. Does anyone
>> know what a value of 1, 2, or 3 means? I tried to find out from the
>> Asterisk source code but it wasn't obvious.
>>
>> Thank you in advance for any tips.
>>
>
> The states you are referring to are chan_sip specific. They are defined
> here[1].
>
> 0 = Disabled
> 1 = We've sent a reinvite to switch to T.38
> 2 = They've sent a reinvite to switch to T38
> 3 = T38 was negotiated and is active
> 4 = T38 negotiation failed/was rejected
>
> [1]
> https://github.com/asterisk/asterisk/blob/master/channels/sip/include/sip.h#L662
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



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[asterisk-users] T38 state values

[David Cunningham]

Hello,

We have a problem where one fax ATA connected to Asterisk works, and
another ATA with the same model and firmware does not. Both are configured
to use T38.

Basically the call comes in to Asterisk which then does a Dial to the ATA
that's registered via SIP, and once that's set up then there's a re-INVITE
from the ATA to switch to T38. The ATA sends a "t30ind: no-signal" after
the 200 OK, and both work up until this point. But then in the working
scenario Asterisk sends a "t30ind: cng" back to the ATA, and in the
not-working scenario Asterisk does not send any T38 data at all.

We've noticed that in the working scenario when the 200 OK is sent from
Asterisk to the ATA, Asterisk logs:

[Mar 17 16:07:50] DEBUG[53838][C-0013e292] chan_sip.c: T38 state changed to
2 on channel SIP/xx.xx.246.70:5060-0030dde1
[Mar 17 16:07:50] DEBUG[6862][C-0013e292] chan_sip.c: T38 state changed to
1 on channel SIP/product-local-bw70-0030ddd9

Whereas in the not working scenario it logs:

[Mar 17 16:23:05] DEBUG[53838][C-0013e39a] chan_sip.c: T38 state changed to
3 on channel SIP/product-local-bw70-0030e0a0
[Mar 17 16:23:05] DEBUG[24973][C-0013e39a] chan_sip.c: T38 state changed to
3 on channel SIP/xx.xx.246.70:5060-0030e0a1

Notice the difference in the "T38 state changed to" values. Does anyone
know what a value of 1, 2, or 3 means? I tried to find out from the
Asterisk source code but it wasn't obvious.

Thank you in advance for any tips.

-- 
David Cunningham, Voisonics Limited
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Re: [asterisk-users] Dahdi fails: fatal error: linux/pci-aspm.h: No such file or directory

[David Cunningham]

Thanks Dovid. Installing from git does indeed work, I was wondering whether
it had been released in a version and if so what version(s) that would be.


On Tue, 22 Feb 2022 at 14:55, Dovid Bender  wrote:

> David,
>
> I vaguely remember having this issue on “newer” versions of Linux. I build
> from git and it works every time. I will try to look at my scripts and post
> later exactly what I do.
>
> On Mon, Feb 21, 2022 at 20:52 David Cunningham 
> wrote:
>
>> Hello,
>>
>> I see some emails about a Dahdi compilation problem with
>> "linux/pci-aspm.h: No such file or directory" two years ago, which suggest
>> trying the "next" branch.
>>
>> Did this change go into a Dahdi release, and if so which version
>> number(s) please?
>>
>> Thank you,
>>
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
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[asterisk-users] Dahdi fails: fatal error: linux/pci-aspm.h: No such file or directory

[David Cunningham]

Hello,

I see some emails about a Dahdi compilation problem with "linux/pci-aspm.h:
No such file or directory" two years ago, which suggest trying the "next"
branch.

Did this change go into a Dahdi release, and if so which version number(s)
please?

Thank you,

-- 
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Re: [asterisk-users] Willing to pay for patch to Asterisk fax detection

[David Cunningham]

Hi Naveen,

Yes indeed, Asterisk does have automatic fax detection. The chan_sip
version just doesn't have all the features we need, namely the ability to
restrict it to the first X seconds of a call, and enabling it on specific
calls only.

A patch to the existing automatic fax detection code to add the features we
need is what we're looking to hire someone for.

Thanks.


On Fri, 12 Nov 2021 at 13:20, David Cunningham 
wrote:

> Hi Antony,
>
> Thanks for the suggestion. I didn't get a response on my request to join
> the asterisk-dev mailing list. I'll try asterisk-biz as well.
>
>
> On Fri, 12 Nov 2021 at 12:23, Antony Stone <
> antony.st...@asterisk.open.source.it> wrote:
>
>> On Thursday 11 November 2021 at 22:29:34, David Cunningham wrote:
>>
>> > Hello,
>> >
>> > We have a commercial client
>>
>> > If anyone has ideas for other places to advertise this request let me
>> know!
>>
>> I would suggest http://lists.digium.com/mailman/listinfo/asterisk-biz
>> because
>> that is the commercial list (you have currently posted to the
>> "non-commercial
>> discussion" list), and
>> http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>>
>> Antony.
>>
>> --
>> "I find the whole business of religion profoundly interesting.  But it
>> does
>> mystify me that otherwise intelligent people take it seriously."
>>
>>  - Douglas Adams
>>
>>Please reply to the
>> list;
>>  please *don't*
>> CC me.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
>


-- 
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-- 
_
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Re: [asterisk-users] Willing to pay for patch to Asterisk fax detection

[David Cunningham]

Hi Antony,

Thanks for the suggestion. I didn't get a response on my request to join
the asterisk-dev mailing list. I'll try asterisk-biz as well.


On Fri, 12 Nov 2021 at 12:23, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Thursday 11 November 2021 at 22:29:34, David Cunningham wrote:
>
> > Hello,
> >
> > We have a commercial client
>
> > If anyone has ideas for other places to advertise this request let me
> know!
>
> I would suggest http://lists.digium.com/mailman/listinfo/asterisk-biz
> because
> that is the commercial list (you have currently posted to the
> "non-commercial
> discussion" list), and
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> Antony.
>
> --
> "I find the whole business of religion profoundly interesting.  But it
> does
> mystify me that otherwise intelligent people take it seriously."
>
>  - Douglas Adams
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



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[asterisk-users] Willing to pay for patch to Asterisk fax detection

[David Cunningham]

Hello,

We have a commercial client who wants automatic fax detection, but the
existing functionality in Asterisk doesn't quite meet their needs. We're
willing to pay for a patch to do the following:

1. Limit the automatic fax detection to the first X seconds of a call. X
could be defined system-wide in sip.conf or in an Asterisk dialplan
variable.

2. Enable or disable fax detection for individual calls. This could be set
with an Asterisk dialplan variable.

3. The patch needs to work with chan_sip on Asterisk 13 and above.

If you're able to help with this please let me know so we can discuss
pricing and your Asterisk development experience.

If anyone has ideas for other places to advertise this request let me know!

Thanks very much,

-- 
David Cunningham, Voisonics Limited
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Re: [asterisk-users] dahdi-linux-complete-3.1.0+3.1.0 : issue on CentOS 7.9 but ot on CentOS 6.10

[David Cunningham]

Hello,

Which version of DAHDI to use depends on the Linux kernel version, as given
by "uname -r". Roughly, Linux kernel 3.x should use DAHDI 2.9, kernel >=
4.0 and < 4.15 should use DAHDI 2.11, and kernel >= 4.15 or greater should
use DAHDI 3.x.

I hope this helps.


On Tue, 23 Feb 2021 at 04:35, Jonas Kellens 
wrote:

> Hello List
>
>
> to answer my own question, and for whom it may interest, I no longer have
> the error about libtonezone.so with Dahdi version :
> dahdi-linux-complete-2.11.1+2.11.1
>
> I don't know what the difference is between Dahdi 2.x and Dahdi 3.x but I
> can say that THERE IS somewhere a difference, as I experienced on Centos
> 7.9.
>
>
>
> Kind regards.
>
>
>
> Op 12-02-21 om 19:11 schreef Jonas Kellens:
>
> Hello list
>
>
> when installing latest DAHDI (dahdi-linux-complete-3.1.0+3.1.0) for usage
> with asterisk-certified-13.21-cert6 on CentOS 6.10 all works well when
> starting dahdi with "/sbin/service dahdi start".
>
>
> But when installing the same DAHDI version in CentOS 7.9 I get the error :*
> /usr/sbin/dahdi_cfg: error while loading shared libraries:
> libtonezone.so.2: cannot open shared object file: No such file or directory*
> when issuing "systemctl start dahdi.service"
>
>
> Is there something missing on my CentOS 7.9 system to work with the latest
> DAHDI version ?
>
> Or is there a better DAHDI version to be used on CentOS 7.9 ?
>
>
> libtonezone is present on my CentOS 7.9 system :
>
> [root@server admin]# locate libtonezone
> /usr/lib/libtonezone.a
> /usr/lib/libtonezone.la
> /usr/lib/libtonezone.so
> /usr/lib/libtonezone.so.1
> /usr/lib/libtonezone.so.1.0
> /usr/lib/libtonezone.so.2
> /usr/lib/libtonezone.so.2.0
> /usr/lib/libtonezone.so.2.0.0
>
>
>
>
> Kind regards.
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



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[asterisk-users] DAHDI compile failed in xusb_libusb.c

[David Cunningham]

Hello,

We have just upgraded a server to Ubuntu 18.04, running kernel
4.15.0-142-generic. Compiling DAHDI 3.0.0 fails with the error below, and
3.1.0 gives the same error. The gcc compiler is version 7.5.0. Would anyone
know what the solution is, or should we open a bug report? Thanks in
advance for any help!

root@ast1:/usr/src/dahdi-linux-complete-3.0.0+3.0.0# make install-config
make -C linux all
make[1]: Entering directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/linux'
make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/firmware'
make[2]: Leaving directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/firmware'
make -C /lib/modules/4.15.0-142-generic/build
SUBDIRS=/usr/src/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi
DAHDI_INCLUDE=/usr/src/dahdi-linux-complete-3.0.0+3.0.0/linux/include
DAHDI_MODULES_EXTRA=" " HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m
make[2]: Entering directory '/usr/src/linux-headers-4.15.0-142-generic'
  Building modules, stage 2.
  MODPOST 28 modules
WARNING: could not find
/usr/src/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd
for
/usr/src/dahdi-linux-complete-3.0.0+3.0.0/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o
make[2]: Leaving directory '/usr/src/linux-headers-4.15.0-142-generic'
make[1]: Leaving directory '/usr/src/dahdi-linux-complete-3.0.0+3.0.0/linux'
(cd tools && autoreconf -i && [ -f config.status ] || ./configure
--with-dahdi=../linux)
make -C tools all
make[1]: Entering directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools'
make  all-recursive
make[2]: Entering directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools'
Making all in xpp
make[3]: Entering directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools/xpp'
Making all in perl_modules
make[4]: Entering directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools/xpp/perl_modules'
make[4]: Nothing to be done for 'all'.
make[4]: Leaving directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools/xpp/perl_modules'
Making all in oct612x
make[4]: Entering directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools/xpp/oct612x'
make[4]: Nothing to be done for 'all'.
make[4]: Leaving directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools/xpp/oct612x'
Making all in xtalk
make[4]: Entering directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools/xpp/xtalk'
  CC   libxtalk_la-xusb_libusb.lo
xusb_libusb.c: In function ‘xusb_find_bypath’:
xusb_libusb.c:500:41: error: ‘/’ directive output may be truncated writing
1 byte into a region of size between 0 and 4093 [-Werror=format-truncation=]
snprintf(devpath_tail, PATH_MAX, "%3s/%3s",
 ^
In file included from /usr/include/stdio.h:862:0,
 from xusb_libusb.c:23:
/usr/include/x86_64-linux-gnu/bits/stdio2.h:64:10: note:
‘__builtin___snprintf_chk’ output between 8 and 8194 bytes into a
destination of size 4096
   return __builtin___snprintf_chk (__s, __n, __USE_FORTIFY_LEVEL - 1,
  ^~~~
__bos (__s), __fmt, __va_arg_pack ());
~
cc1: all warnings being treated as errors
Makefile:658: recipe for target 'libxtalk_la-xusb_libusb.lo' failed
make[4]: *** [libxtalk_la-xusb_libusb.lo] Error 1
make[4]: Leaving directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools/xpp/xtalk'
Makefile:1043: recipe for target 'all-recursive' failed
make[3]: *** [all-recursive] Error 1
make[3]: Leaving directory
'/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools/xpp'
Makefile:1115: recipe for target 'all-recursive' failed
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory '/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools'
Makefile:664: recipe for target 'all' failed
make[1]: *** [all] Error 2
make[1]: Leaving directory '/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools'
Makefile:9: recipe for target 'all' failed
make: *** [all] Error 2


-- 
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Re: [asterisk-users] faxdetect timeout configuration

[David Cunningham]

Hi Steve,

Thanks for that. Perhaps the change to res_fax might help us? I'm hoping
someone can say whether or not for sure.


On Wed, 30 Dec 2020 at 11:00, Steve Edwards 
wrote:

> On Wed, 30 Dec 2020, David Cunningham wrote:
>
> > Would anyone be able to tell us how to configure this option for calls
> > arriving via chan_sip?
>
> A 30,000 ft peek suggests you're out of luck unless you switch to pjsip:
>
> -ws10::sedwards:~$ rgrep -l faxdetect_timeout /usr/src/asterisk-17.4.0/
> /usr/src/asterisk-17.4.0/CHANGES
> /usr/src/asterisk-17.4.0/ChangeLog
> /usr/src/asterisk-17.4.0/channels/chan_dahdi.c
> /usr/src/asterisk-17.4.0/channels/chan_dahdi.h
> /usr/src/asterisk-17.4.0/channels/chan_misdn.c
> /usr/src/asterisk-17.4.0/channels/chan_pjsip.c
> /usr/src/asterisk-17.4.0/channels/misdn/chan_misdn_config.h
> /usr/src/asterisk-17.4.0/channels/misdn_config.c
> /usr/src/asterisk-17.4.0/configs/samples/chan_dahdi.conf.sample
> /usr/src/asterisk-17.4.0/include/asterisk/res_fax.h
> /usr/src/asterisk-17.4.0/include/asterisk/res_pjsip.h
> /usr/src/asterisk-17.4.0/res/res_fax.c
> /usr/src/asterisk-17.4.0/res/res_pjsip/pjsip_configuration.c
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>  https://www.linkedin.com/in/steve-edwards-4244281
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users



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[asterisk-users] faxdetect timeout configuration

[David Cunningham]

Hello,

We see there is addition of a faxdetect_timeout option and fix for
FAXOPT(faxdetect) noted in the Asterisk 13 change log:

https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/asterisk-certified-13.8-cert2-summary.html

Would anyone be able to tell us how to configure this option for calls
arriving via chan_sip? Is it just a matter of setting the FAXOPT(faxdetect)
variable in the dialplan? What we'd like to do is restrict fax detection to
the first N seconds of a call.

Thanks very much for any advice,

-- 
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New Zealand: +64 (0)28 2558 3782
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Re: [asterisk-users] NAT problem with recvonly calls

[David Cunningham]

Hi Dovid,

Thanks for that. Can you explain how the Progress() and/or Playback()
actually help the NAT problem? I'm trying to figure out how it tells
Asterisk the correct address to send the RTP to.


On Thu, 3 Dec 2020 at 16:10, Dovid Bender  wrote:

> David,
>
> You should be able to do that via the agi as well.
>
> On Wed, Dec 2, 2020 at 20:32 David Cunningham 
> wrote:
>
>> Hi Dovid,
>>
>> We're using Enswitch so it uses AGI rather than a regular Asterisk
>> dialplan, but perhaps sending it to a custom-made Asterisk context with the
>> lines you suggest could be the best way forward.
>>
>> Thank you for that.
>>
>>
>> On Thu, 3 Dec 2020 at 13:01, Dovid Bender  wrote:
>>
>>> David,
>>>
>>> Does Asterisk send a 180 or a 183 with SDP? We have found that using
>>> these two lines help (where xc is a 500ms blank sound file)
>>> Exten => _X.,n, Progress()
>>> Exten => _X.,n, Playback(xc,noanswer)
>>>
>>>
>>> On Wed, Dec 2, 2020 at 4:30 PM David Cunningham <
>>> dcunning...@voisonics.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> We have a  problem with a SIP doorbell device which sends media one way
>>>> only, and NAT at the receiving device.
>>>>
>>>> When the doorbell button is pressed it makes a call to a configured
>>>> destination. Since the doorbell only sends and doesn't receive it sends the
>>>> INVITE with sendonly in the SDP, and the destination then replies with a
>>>> 200 OK with recvonly in the SDP.
>>>>
>>>> The problem is that the destination is behind NAT, and its reply
>>>> contains a private network IP in the SDP. Normally Asterisk when nat=yes
>>>> works around that by adjusting the destination for RTP to be the address it
>>>> actually receives audio from, however because this device is recvonly
>>>> Asterisk never receives audio from it. This means Asterisk keeps trying to
>>>> send the doorbell's RTP to the private network IP which of course fails,
>>>> and the destination never gets the RTP from the doorbell.
>>>>
>>>> Does anyone know how to work around this issue?
>>>>
>>>> Thank you in advance,
>>>>
>>>> --
>>>> David Cunningham, Voisonics Limited
>>>> http://voisonics.com/
>>>> USA: +1 213 221 1092
>>>> New Zealand: +64 (0)28 2558 3782
>>>> --
>>>> _
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>
>>>> Check out the new Asterisk community forum at:
>>>> https://community.asterisk.org/
>>>>
>>>> New to Asterisk? Start here:
>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
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Re: [asterisk-users] NAT problem with recvonly calls

[David Cunningham]

Hi Dovid,

We're using Enswitch so it uses AGI rather than a regular Asterisk
dialplan, but perhaps sending it to a custom-made Asterisk context with the
lines you suggest could be the best way forward.

Thank you for that.


On Thu, 3 Dec 2020 at 13:01, Dovid Bender  wrote:

> David,
>
> Does Asterisk send a 180 or a 183 with SDP? We have found that using these
> two lines help (where xc is a 500ms blank sound file)
> Exten => _X.,n, Progress()
> Exten => _X.,n, Playback(xc,noanswer)
>
>
> On Wed, Dec 2, 2020 at 4:30 PM David Cunningham 
> wrote:
>
>> Hello,
>>
>> We have a  problem with a SIP doorbell device which sends media one way
>> only, and NAT at the receiving device.
>>
>> When the doorbell button is pressed it makes a call to a configured
>> destination. Since the doorbell only sends and doesn't receive it sends the
>> INVITE with sendonly in the SDP, and the destination then replies with a
>> 200 OK with recvonly in the SDP.
>>
>> The problem is that the destination is behind NAT, and its reply contains
>> a private network IP in the SDP. Normally Asterisk when nat=yes works
>> around that by adjusting the destination for RTP to be the address it
>> actually receives audio from, however because this device is recvonly
>> Asterisk never receives audio from it. This means Asterisk keeps trying to
>> send the doorbell's RTP to the private network IP which of course fails,
>> and the destination never gets the RTP from the doorbell.
>>
>> Does anyone know how to work around this issue?
>>
>> Thank you in advance,
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



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[asterisk-users] NAT problem with recvonly calls

[David Cunningham]

Hello,

We have a  problem with a SIP doorbell device which sends media one way
only, and NAT at the receiving device.

When the doorbell button is pressed it makes a call to a configured
destination. Since the doorbell only sends and doesn't receive it sends the
INVITE with sendonly in the SDP, and the destination then replies with a
200 OK with recvonly in the SDP.

The problem is that the destination is behind NAT, and its reply contains a
private network IP in the SDP. Normally Asterisk when nat=yes works around
that by adjusting the destination for RTP to be the address it actually
receives audio from, however because this device is recvonly Asterisk never
receives audio from it. This means Asterisk keeps trying to send the
doorbell's RTP to the private network IP which of course fails, and the
destination never gets the RTP from the doorbell.

Does anyone know how to work around this issue?

Thank you in advance,

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
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Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[David Cunningham]

Thanks for the suggestions. We'd prefer not to complicate the architecture
with additional proxies in front, so will try setting the Linux network
routes to see if that helps.


On Fri, 30 Oct 2020 at 16:24, John Runyon  wrote:

> David, can you play around with the routing table and get the OS to handle
> it for you? So long as asterisk isn’t calling bind() (or is calling with
> 0.0.0.0) I would imagine adding a route for the peer, with your normal
> gateway, and the correct device would work.
>
> On Thu, Oct 29, 2020 at 10:04 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Dovid,
>>
>> We can change the SDP in Kamailio, but Asterisk will still send its RTP
>> from its default address. The remote end is strict about accepting RTP from
>> the specified source and won't accept it. Have you any suggestions to solve
>> that problem?
>>
>> Thank you.
>>
>>
>> On Fri, 30 Oct 2020 at 14:49, Dovid Bender  wrote:
>>
>>> Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you
>>> pass it along as is. Where you want 2.2.2.2 change the sdp in
>>> opensips/kamailio
>>>
>>> On Thu, Oct 29, 2020 at 20:44 David Cunningham <
>>> dcunning...@voisonics.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> Does anyone know a way with chan_sip to tell Asterisk to use a specific
>>>> IP address for its end of the communication for a specific device?
>>>> Something like:
>>>>
>>>> [device]
>>>> type = friend
>>>> host = 11.22.11.22
>>>> ouraddress = 33.44.33.44
>>>>
>>>> This is for use on a server with multiple IP addresses. There is the
>>>> "extenip" setting, but it's really designed for NAT, and can only appear in
>>>> the [general] section.
>>>>
>>>> Any suggestions would be greatly appreciated.
>>>>
>>>>
>>>> On Sat, 24 Oct 2020 at 09:43, David Cunningham <
>>>> dcunning...@voisonics.com> wrote:
>>>>
>>>>> OK, thank you George.
>>>>>
>>>>>
>>>>> On Sat, 24 Oct 2020 at 03:16, George Joseph 
>>>>> wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <
>>>>>> dcunning...@voisonics.com> wrote:
>>>>>>
>>>>>>> Hi George,
>>>>>>>
>>>>>>> Thank you for the response. I'm a little unclear on what you mean by
>>>>>>> a transport. We're using chan_sip, not pjsip.
>>>>>>>
>>>>>>> Do you mean a device in sip.conf, using bindaddr to set the address
>>>>>>> to bind for that device? We've only used bindaddr in the [general] 
>>>>>>> section
>>>>>>> before, but if it will work in a device that could be the answer.
>>>>>>>
>>>>>>
>>>>>> Sorry.  I just assume chan_pjsip these days.  Not sure how you'd do
>>>>>> it for chan_sip.
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Fri, 23 Oct 2020 at 00:13, George Joseph 
>>>>>>> wrote:
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
>>>>>>>> dcunning...@voisonics.com> wrote:
>>>>>>>>
>>>>>>>>> Hello,
>>>>>>>>>
>>>>>>>>> We have an Asterisk server with two public IP addresses, let's say
>>>>>>>>> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are 
>>>>>>>>> bridged with
>>>>>>>>> a call dialled from Asterisk to an external destination. The external
>>>>>>>>> destination sees the SIP packet as coming from 1.1.1.1 and the media
>>>>>>>>> address in the SDP is 1.1.1.1, which is great.
>>>>>>>>>
>>>>>>>>> However if we receive a call in to 2.2.2.2 then the call dialled
>>>>>>>>> from Asterisk to an external destination still comes from 1.1.1.1, 
>>>>>>>>> whereas
>>>>>>>>> we want it to come from 2.2.2.2. 

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[David Cunningham]

Hi Dovid,

We can change the SDP in Kamailio, but Asterisk will still send its RTP
from its default address. The remote end is strict about accepting RTP from
the specified source and won't accept it. Have you any suggestions to solve
that problem?

Thank you.


On Fri, 30 Oct 2020 at 14:49, Dovid Bender  wrote:

> Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass
> it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio
>
> On Thu, Oct 29, 2020 at 20:44 David Cunningham 
> wrote:
>
>> Hello,
>>
>> Does anyone know a way with chan_sip to tell Asterisk to use a specific
>> IP address for its end of the communication for a specific device?
>> Something like:
>>
>> [device]
>> type = friend
>> host = 11.22.11.22
>> ouraddress = 33.44.33.44
>>
>> This is for use on a server with multiple IP addresses. There is the
>> "extenip" setting, but it's really designed for NAT, and can only appear in
>> the [general] section.
>>
>> Any suggestions would be greatly appreciated.
>>
>>
>> On Sat, 24 Oct 2020 at 09:43, David Cunningham 
>> wrote:
>>
>>> OK, thank you George.
>>>
>>>
>>> On Sat, 24 Oct 2020 at 03:16, George Joseph  wrote:
>>>
>>>>
>>>>
>>>> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <
>>>> dcunning...@voisonics.com> wrote:
>>>>
>>>>> Hi George,
>>>>>
>>>>> Thank you for the response. I'm a little unclear on what you mean by a
>>>>> transport. We're using chan_sip, not pjsip.
>>>>>
>>>>> Do you mean a device in sip.conf, using bindaddr to set the address to
>>>>> bind for that device? We've only used bindaddr in the [general] section
>>>>> before, but if it will work in a device that could be the answer.
>>>>>
>>>>
>>>> Sorry.  I just assume chan_pjsip these days.  Not sure how you'd do it
>>>> for chan_sip.
>>>>
>>>>
>>>>
>>>>>
>>>>>
>>>>> On Fri, 23 Oct 2020 at 00:13, George Joseph 
>>>>> wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
>>>>>> dcunning...@voisonics.com> wrote:
>>>>>>
>>>>>>> Hello,
>>>>>>>
>>>>>>> We have an Asterisk server with two public IP addresses, let's say
>>>>>>> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged 
>>>>>>> with
>>>>>>> a call dialled from Asterisk to an external destination. The external
>>>>>>> destination sees the SIP packet as coming from 1.1.1.1 and the media
>>>>>>> address in the SDP is 1.1.1.1, which is great.
>>>>>>>
>>>>>>> However if we receive a call in to 2.2.2.2 then the call dialled
>>>>>>> from Asterisk to an external destination still comes from 1.1.1.1, 
>>>>>>> whereas
>>>>>>> we want it to come from 2.2.2.2. The source of any dialled call (the IP
>>>>>>> packet and the SDP media address) should be the same as the address the
>>>>>>> related inbound call was received to.
>>>>>>>
>>>>>>> For example:
>>>>>>> INVITE received to 1.1.1.1:5060 -> Asterisk dials
>>>>>>> destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
>>>>>>> termination.com
>>>>>>> INVITE received to 2.2.2.2:5060 -> Asterisk dials
>>>>>>> destinat...@pstn.com -> INVITE sent from 2.2.2.2:5060 to pstn.com
>>>>>>>
>>>>>>> Does anyone know how this can be achieved?
>>>>>>>
>>>>>>
>>>>>> If termination.com is only on 1.1.1.1 and pstn.com is only on
>>>>>> 2.2.2.2, create 2 transports, one specifically bound to 1.1.1.1,
>>>>>> transport-1.1.1.1 for instance, and another to 2.2.2.2:
>>>>>> transport-2.2.2.2.  The names aren't important as long as you can tell 
>>>>>> the
>>>>>> difference.  Then explicitly configure endpoint termination.com's
>>>>>> "transport" parameter to "transport-1.1.1.1" and pstn.com's
>>>>>> "transp

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[David Cunningham]

Hello,

Does anyone know a way with chan_sip to tell Asterisk to use a specific IP
address for its end of the communication for a specific device? Something
like:

[device]
type = friend
host = 11.22.11.22
ouraddress = 33.44.33.44

This is for use on a server with multiple IP addresses. There is the
"extenip" setting, but it's really designed for NAT, and can only appear in
the [general] section.

Any suggestions would be greatly appreciated.


On Sat, 24 Oct 2020 at 09:43, David Cunningham 
wrote:

> OK, thank you George.
>
>
> On Sat, 24 Oct 2020 at 03:16, George Joseph  wrote:
>
>>
>>
>> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <
>> dcunning...@voisonics.com> wrote:
>>
>>> Hi George,
>>>
>>> Thank you for the response. I'm a little unclear on what you mean by a
>>> transport. We're using chan_sip, not pjsip.
>>>
>>> Do you mean a device in sip.conf, using bindaddr to set the address to
>>> bind for that device? We've only used bindaddr in the [general] section
>>> before, but if it will work in a device that could be the answer.
>>>
>>
>> Sorry.  I just assume chan_pjsip these days.  Not sure how you'd do it
>> for chan_sip.
>>
>>
>>
>>>
>>>
>>> On Fri, 23 Oct 2020 at 00:13, George Joseph  wrote:
>>>
>>>>
>>>>
>>>> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
>>>> dcunning...@voisonics.com> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> We have an Asterisk server with two public IP addresses, let's say
>>>>> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged 
>>>>> with
>>>>> a call dialled from Asterisk to an external destination. The external
>>>>> destination sees the SIP packet as coming from 1.1.1.1 and the media
>>>>> address in the SDP is 1.1.1.1, which is great.
>>>>>
>>>>> However if we receive a call in to 2.2.2.2 then the call dialled from
>>>>> Asterisk to an external destination still comes from 1.1.1.1, whereas we
>>>>> want it to come from 2.2.2.2. The source of any dialled call (the IP 
>>>>> packet
>>>>> and the SDP media address) should be the same as the address the related
>>>>> inbound call was received to.
>>>>>
>>>>> For example:
>>>>> INVITE received to 1.1.1.1:5060 -> Asterisk dials
>>>>> destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
>>>>> termination.com
>>>>> INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@pstn.com
>>>>> -> INVITE sent from 2.2.2.2:5060 to pstn.com
>>>>>
>>>>> Does anyone know how this can be achieved?
>>>>>
>>>>
>>>> If termination.com is only on 1.1.1.1 and pstn.com is only on 2.2.2.2,
>>>> create 2 transports, one specifically bound to 1.1.1.1, transport-1.1.1.1
>>>> for instance, and another to 2.2.2.2:  transport-2.2.2.2.  The names
>>>> aren't important as long as you can tell the difference.  Then explicitly
>>>> configure endpoint termination.com's "transport" parameter to
>>>> "transport-1.1.1.1" and pstn.com's "transport" parameter to
>>>> "transport-2.2.2.2".   In your dialplan, you can see which endpoint the
>>>> call came in on, and route it out the same endpoint.
>>>>
>>>> If both providers are available from both interfaces, you can create 2
>>>> endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
>>>> termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with the
>>>> same transports as above.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>>
>>>>> Thanks in advance for your help,
>>>>>
>>>>> --
>>>>> David Cunningham, Voisonics Limited
>>>>> http://voisonics.com/
>>>>> USA: +1 213 221 1092
>>>>> New Zealand: +64 (0)28 2558 3782
>>>>> --
>>>>> _
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>>
>>>>> Check out the new Asterisk community forum at:
>>>>> https://community.asterisk.org/
>>>>>
>>>>> New to Asterisk? 

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[David Cunningham]

OK, thank you George.


On Sat, 24 Oct 2020 at 03:16, George Joseph  wrote:

>
>
> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi George,
>>
>> Thank you for the response. I'm a little unclear on what you mean by a
>> transport. We're using chan_sip, not pjsip.
>>
>> Do you mean a device in sip.conf, using bindaddr to set the address to
>> bind for that device? We've only used bindaddr in the [general] section
>> before, but if it will work in a device that could be the answer.
>>
>
> Sorry.  I just assume chan_pjsip these days.  Not sure how you'd do it for
> chan_sip.
>
>
>
>>
>>
>> On Fri, 23 Oct 2020 at 00:13, George Joseph  wrote:
>>
>>>
>>>
>>> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
>>> dcunning...@voisonics.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> We have an Asterisk server with two public IP addresses, let's say
>>>> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with
>>>> a call dialled from Asterisk to an external destination. The external
>>>> destination sees the SIP packet as coming from 1.1.1.1 and the media
>>>> address in the SDP is 1.1.1.1, which is great.
>>>>
>>>> However if we receive a call in to 2.2.2.2 then the call dialled from
>>>> Asterisk to an external destination still comes from 1.1.1.1, whereas we
>>>> want it to come from 2.2.2.2. The source of any dialled call (the IP packet
>>>> and the SDP media address) should be the same as the address the related
>>>> inbound call was received to.
>>>>
>>>> For example:
>>>> INVITE received to 1.1.1.1:5060 -> Asterisk dials
>>>> destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
>>>> termination.com
>>>> INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@pstn.com
>>>> -> INVITE sent from 2.2.2.2:5060 to pstn.com
>>>>
>>>> Does anyone know how this can be achieved?
>>>>
>>>
>>> If termination.com is only on 1.1.1.1 and pstn.com is only on 2.2.2.2,
>>> create 2 transports, one specifically bound to 1.1.1.1, transport-1.1.1.1
>>> for instance, and another to 2.2.2.2:  transport-2.2.2.2.  The names
>>> aren't important as long as you can tell the difference.  Then explicitly
>>> configure endpoint termination.com's "transport" parameter to
>>> "transport-1.1.1.1" and pstn.com's "transport" parameter to
>>> "transport-2.2.2.2".   In your dialplan, you can see which endpoint the
>>> call came in on, and route it out the same endpoint.
>>>
>>> If both providers are available from both interfaces, you can create 2
>>> endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
>>> termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with the
>>> same transports as above.
>>>
>>>
>>>
>>>
>>>
>>>>
>>>> Thanks in advance for your help,
>>>>
>>>> --
>>>> David Cunningham, Voisonics Limited
>>>> http://voisonics.com/
>>>> USA: +1 213 221 1092
>>>> New Zealand: +64 (0)28 2558 3782
>>>> --
>>>> _
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>
>>>> Check out the new Asterisk community forum at:
>>>> https://community.asterisk.org/
>>>>
>>>> New to Asterisk? Start here:
>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>>> --
>>> George Joseph
>>> Asterisk Software Developer
>>> direct/fax +1 256 428 6012
>>> Check us out at www.sangoma.com and www.asterisk.org
>>> [image: image.png]
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wi

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[David Cunningham]

Hi George,

Thank you for the response. I'm a little unclear on what you mean by a
transport. We're using chan_sip, not pjsip.

Do you mean a device in sip.conf, using bindaddr to set the address to bind
for that device? We've only used bindaddr in the [general] section before,
but if it will work in a device that could be the answer.


On Fri, 23 Oct 2020 at 00:13, George Joseph  wrote:

>
>
> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>> We have an Asterisk server with two public IP addresses, let's say
>> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with
>> a call dialled from Asterisk to an external destination. The external
>> destination sees the SIP packet as coming from 1.1.1.1 and the media
>> address in the SDP is 1.1.1.1, which is great.
>>
>> However if we receive a call in to 2.2.2.2 then the call dialled from
>> Asterisk to an external destination still comes from 1.1.1.1, whereas we
>> want it to come from 2.2.2.2. The source of any dialled call (the IP packet
>> and the SDP media address) should be the same as the address the related
>> inbound call was received to.
>>
>> For example:
>> INVITE received to 1.1.1.1:5060 -> Asterisk dials
>> destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
>> termination.com
>> INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@pstn.com
>> -> INVITE sent from 2.2.2.2:5060 to pstn.com
>>
>> Does anyone know how this can be achieved?
>>
>
> If termination.com is only on 1.1.1.1 and pstn.com is only on 2.2.2.2,
> create 2 transports, one specifically bound to 1.1.1.1, transport-1.1.1.1
> for instance, and another to 2.2.2.2:  transport-2.2.2.2.  The names
> aren't important as long as you can tell the difference.  Then explicitly
> configure endpoint termination.com's "transport" parameter to
> "transport-1.1.1.1" and pstn.com's "transport" parameter to
> "transport-2.2.2.2".   In your dialplan, you can see which endpoint the
> call came in on, and route it out the same endpoint.
>
> If both providers are available from both interfaces, you can create 2
> endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
> termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with the
> same transports as above.
>
>
>
>
>
>>
>> Thanks in advance for your help,
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> George Joseph
> Asterisk Software Developer
> direct/fax +1 256 428 6012
> Check us out at www.sangoma.com and www.asterisk.org
> [image: image.png]
> --
> _____
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[David Cunningham]

Hello,

We have an Asterisk server with two public IP addresses, let's say 1.1.1.1
and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call
dialled from Asterisk to an external destination. The external destination
sees the SIP packet as coming from 1.1.1.1 and the media address in the SDP
is 1.1.1.1, which is great.

However if we receive a call in to 2.2.2.2 then the call dialled from
Asterisk to an external destination still comes from 1.1.1.1, whereas we
want it to come from 2.2.2.2. The source of any dialled call (the IP packet
and the SDP media address) should be the same as the address the related
inbound call was received to.

For example:
INVITE received to 1.1.1.1:5060 -> Asterisk dials
destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
termination.com
INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@pstn.com ->
INVITE sent from 2.2.2.2:5060 to pstn.com

Does anyone know how this can be achieved?

Thanks in advance for your help,

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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[asterisk-users] PJSIP - Forcing codec preference?

[David Herselman]

Hi,

We're holding ourselves back from moving to PJSIP as we don't appear to have 
figured out how to force codec preference in a dial plan. The 'PJSIP Advanced 
Codec Negotiation' document 
(https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation) 
appears to ultimately be what we're after, but we're not comfortable running 
Asterisk 18 in production just yet. Is there no way to mimic functionality we 
previously had in chan_sip?


I don't appear to be able to set an inheritable variable for the subsequent 
PJSIP leg of the call, to exclusively only offer the codec we negotiated for 
the first leg of the call. If for example we have chan_iax2 incoming that we 
wish to send out via pjsip.

With chan_sip, this works:
 exten => s,n,Set(_SIP_CODEC_OUTBOUND=${CHANNEL(audioreadformat)})

With pjsip, this gives an error:
  exten => s,n,Set(_PJSIP_MEDIA_OFFER(audio)=!all,${CHANNEL(audioreadformat)})

Error:
  ERROR[26925][C-00020b9c] pbx_functions.c: Function _PJSIP_MEDIA_OFFER not 
registered

I'd image things haven't changed since 2018 where this appears to have been 
discussed in the following thread:
  Re: Pjsip migration - SIP_CODEC and SIP_CODEC_OUTBOUND
  URL: 
https://community.asterisk.org/t/pjsip-migration-sip-codec-and-sip-codec-outbound/73342/7

No way besides learning whatever code the AGI's written in, temporarily passing 
it up as a temporary variable and then calling a pre-dial handler?



I presume incoming shouldn't have this problem as the channel would be pjsip, 
right? With chan_sip we simply set SIP_CODEC as one of the first inbound 
context dial plan rules, before the channel is answered:
  exten => s,n,Set(SIP_CODEC=alaw)

I presume the following would work with pjsip:
  exten => s,n,Set(PJSIP_MEDIA_OFFER(audio)=!all,alaw)

PS: I can't find the reference again but recall a recommendation to call 
Progress() due to nuances with some systems, is this still relevant?


Regards
David Herselman
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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Re: [asterisk-users] Negotiates g729 but RTP contains g711

[David Herselman]

ca3-4db8-81e0-f677e37c24e9>
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] bridge_channel.c: Channel 
SIP/Upstream-00021a0d joined 'simple_bridge' basic-bridge 
<0d377050-bca3-4db8-81e0-f677e37c24e9>
[2020-09-19 23:42:22] VERBOSE[2637] chan_sip.c:

[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020640, ts 000160, len 000160)
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020641, ts 000320, len 000160)
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020642, ts 000480, len 000160)
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020643, ts 000640, len 000160)
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020644, ts 000800, len 000160)
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020645, ts 000960, len 000160)


Regards
David Herselman


From: asterisk-users 
mailto:asterisk-users-boun...@lists.digium.com>>
 On Behalf Of David Herselman
Sent: Wednesday, 23 September 2020 4:17 PM
To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Subject: [asterisk-users] Negotiates g729 but RTP contains g711

Hi,

We have a scenario where inbound calls from an upstream provider (chan_sip) 
sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. 
Both the upstream and downstream trunks are limited to only offering g729 
whilst the initial invite from our upstream provider offers both g711 and g729. 
Asterisk presumably simply forwards the media from iax2 trunk encapsulation to 
sip encapsulation. Most calls surprisingly work, presumably by the caller's 
system identifying the incoming media as g711, whilst very few callers don't 
hear the IVR prompt. The downstream is unfortunately not within our control but 
can't be anything other than Asterisk, considering it's using iax2 in trunk 
mode.

We are running Asterisk 16.13.0, not sure what version the downstream is using.

caller -> upstream -> us -> downstream (IVR)

Herewith the SIP portion of the call, between upstream and us:
Available here: https://ibb.co/jRGvvVc


Wireshark unfortunately still cannot dissect iax2 trunk captures though, so I 
didn't know how to conclusively identify where this problem originates. I do 
however have a concern that the media we are receiving's packet size (74 bytes) 
indicates that it is most likely G729.

Herewith the IAX2 trunk portion of the call, between us and downstream:
Available here: https://ibb.co/r07PkkK


ie: We appear to have a reproducible environment where an inbound SIP trunk 
call sent to a downstream IAX2 trunk negotiates g729 in all 4 streams, receives 
g729 media from downstream iax2 trunk but then transmits g711a upstream.

I'm however struggling with the downstream pcap, to establish what's different 
about these calls. Trunk config and forwarding structure works the identical 
way for 50+ other flows on the same host.


Regards
David Herselman
-- 
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Re: [asterisk-users] Negotiates g729 but RTP contains g711

[David Herselman]

e 8, seq 020999, ts 250248, len 000160)
[2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 021000, ts 250408, len 000160)
[2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 021001, ts 250568, len 000160)
[2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 021002, ts 250728, len 000160)
[2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 021003, ts 250888, len 000160)
[2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 021004, ts 251048, len 000160)
[2020-09-19 23:42:29] VERBOSE[2637] chan_sip.c:
<--- SIP read from UDP:41.11.11.12:5060 --->
BYE sip:01@52.22.22.22:5160 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK5df7.1435d67.0
Via: SIP/2.0/UDP 
41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK577fb6bb;rport=5070
From: "+278" ;tag=as40fe2614
To: ;tag=as11a1cd82
Call-ID: 7030be5a09d89a9543234da051897a49@41.11.11.11
CSeq: 103 BYE
User-Agent: PortaOne
Max-Forwards: 69
Reason: Q.850 ;cause=16; text="Normal Clearing"
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<->
[2020-09-19 23:42:29] VERBOSE[2637] chan_sip.c: --- (13 headers 0 lines) ---
[2020-09-19 23:42:29] VERBOSE[2637][C-00021a1f] chan_sip.c: Sending to 
41.11.11.12:5060 (NAT)
[2020-09-19 23:42:29] VERBOSE[2637][C-00021a1f] chan_sip.c: Scheduling 
destruction of SIP dialog '7030be5a09d89a9543234da051897a49@41.11.11.11' in 
32000 ms (Method: BYE)
[2020-09-19 23:42:29] VERBOSE[2637][C-00021a1f] chan_sip.c:
<--- Transmitting (NAT) to 41.11.11.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
41.11.11.12:5060;branch=z9hG4bK5df7.1435d67.0;received=41.11.11.12;rport=5060
Via: SIP/2.0/UDP 
41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK577fb6bb;rport=5070
Record-Route: 
From: "+278" ;tag=as40fe2614
To: ;tag=as11a1cd82
Call-ID: 7030be5a09d89a9543234da051897a49@41.11.11.11
CSeq: 103 BYE
Server: Asterisk PBX 16.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<>
[2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] bridge_channel.c: Channel 
SIP/Upstream-00021a0d left 'simple_bridge' basic-bridge 
<0d377050-bca3-4db8-81e0-f677e37c24e9>
[2020-09-19 23:42:29] VERBOSE[15154][C-00021a1f] bridge_channel.c: Channel 
IAX2/Downstream-26055 left 'simple_bridge' basic-bridge 
<0d377050-bca3-4db8-81e0-f677e37c24e9>
[2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] pbx.c: Spawn extension 
(incoming, Downstream_01, 1) exited non-zero on 'SIP/Upstream-00021a0d'
[2020-09-19 23:42:29] VERBOSE[15154][C-00021a1f] chan_iax2.c: Hungup 
'IAX2/Downstream-26055'


Regards
David Herselman


From: asterisk-users  On Behalf Of 
David Herselman
Sent: Wednesday, 23 September 2020 4:17 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Negotiates g729 but RTP contains g711

Hi,

We have a scenario where inbound calls from an upstream provider (chan_sip) 
sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. 
Both the upstream and downstream trunks are limited to only offering g729 
whilst the initial invite from our upstream provider offers both g711 and g729. 
Asterisk presumably simply forwards the media from iax2 trunk encapsulation to 
sip encapsulation. Most calls surprisingly work, presumably by the caller's 
system identifying the incoming media as g711, whilst very few callers don't 
hear the IVR prompt. The downstream is unfortunately not within our control but 
can't be anything other than Asterisk, considering it's using iax2 in trunk 
mode.

We are running Asterisk 16.13.0, not sure what version the downstream is using.

caller -> upstream -> us -> downstream (IVR)

Herewith the SIP portion of the call, between upstream and us:
Available here: https://ibb.co/jRGvvVc


Wireshark unfortunately still cannot dissect iax2 trunk captures though, so I 
didn't know how to conclusively identify where this problem originates. I do 
however have a concern that the media we are receiving's packet size (74 bytes) 
indicates that it is most likely G729.

Herewith the IAX2 trunk portion of the call, between us and downstream:
Available here: https://ibb.co/r07PkkK


ie: We appear to have a reproducible environment where an inbound SIP trunk 
call sent to a downstream IAX2 trunk negotiates g729 in all 4 streams, receives 
g729 media from downstream iax2 trunk but then transmits g711a upstream.

I'm however struggling with the downstream pcap, to establish what's different 
about these calls. Trunk config and forwarding structure works the identical

Re: [asterisk-users] Negotiates g729 but RTP contains g711

[David Herselman]

ca3-4db8-81e0-f677e37c24e9>
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] bridge_channel.c: Channel 
SIP/Upstream-00021a0d joined 'simple_bridge' basic-bridge 
<0d377050-bca3-4db8-81e0-f677e37c24e9>
[2020-09-19 23:42:22] VERBOSE[2637] chan_sip.c:

[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020640, ts 000160, len 000160)
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020641, ts 000320, len 000160)
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020642, ts 000480, len 000160)
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020643, ts 000640, len 000160)
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020644, ts 000800, len 000160)
[2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP 
packet to  41.11.11.11:13918 (type 8, seq 020645, ts 000960, len 000160)


Regards
David Herselman
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[asterisk-users] Negotiates g729 but RTP contains g711

[David Herselman]

Hi,

We have a scenario where inbound calls from an upstream provider (chan_sip) 
sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. 
Both the upstream and downstream trunks are limited to only offering g729 
whilst the initial invite from our upstream provider offers both g711 and g729. 
Asterisk presumably simply forwards the media from iax2 trunk encapsulation to 
sip encapsulation. Most calls surprisingly work, presumably by the caller's 
system identifying the incoming media as g711, whilst very few callers don't 
hear the IVR prompt. The downstream is unfortunately not within our control but 
can't be anything other than Asterisk, considering it's using iax2 in trunk 
mode.

We are running Asterisk 16.13.0, not sure what version the downstream is using.

caller -> upstream -> us -> downstream (IVR)

Herewith the SIP portion of the call, between upstream and us:
Available here: https://ibb.co/jRGvvVc


Wireshark unfortunately still cannot dissect iax2 trunk captures though, so I 
didn't know how to conclusively identify where this problem originates. I do 
however have a concern that the media we are receiving's packet size (74 bytes) 
indicates that it is most likely G729.

Herewith the IAX2 trunk portion of the call, between us and downstream:
Available here: https://ibb.co/r07PkkK


ie: We appear to have a reproducible environment where an inbound SIP trunk 
call sent to a downstream IAX2 trunk negotiates g729 in all 4 streams, receives 
g729 media from downstream iax2 trunk but then transmits g711a upstream.

I'm however struggling with the downstream pcap, to establish what's different 
about these calls. Trunk config and forwarding structure works the identical 
way for 50+ other flows on the same host.


Regards
David Herselman
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Re: [asterisk-users] rotatestrategy = none not working

[David Cunningham]

Hi Steve,

Thanks for the answer. Since that's what we already have configured, any
idea why it wouldn't work? As I said, when "asterisk -rx 'logger reload'"
is run it still rotates the log file.


On Wed, 20 May 2020 at 18:37, Steve Edwards 
wrote:

> On Wed, 20 May 2020, David Cunningham wrote:
>
> > We have an Asterisk 11.3 server where we want log rotation handled
> > purely by Linux's logrotate, and not by Asterisk. To this end we've
> > configured the [general] action of /etc/asterisk/logger.conf with:
> >
> > rotatestrategy = none
> >
> > However, an "asterisk -rx 'logger reload'" still rotates the log files.
> > Does anyone know why?
>
> I had to hunt, but I found an 11.17.1 system :)
>
> 'none' does not rotate a log file on this host. Here's my logger.conf:
>
> ; Created by makefile on 2020-05-19 at 23:05:08
> ; from /source/src/obl-server/logger.conf.pre
>
> [general]
>  rotatestrategy  = none
>
> [logfiles]
>  /tmp/ast-log-test   =
> debug,dtmf,error,event,notice,verbose,warning
>
> ; (end of /etc/asterisk/obl/logger.conf)
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>  https://www.linkedin.com/in/steve-edwards-4244281
>
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>
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>http://lists.digium.com/mailman/listinfo/asterisk-users



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[asterisk-users] rotatestrategy = none not working

[David Cunningham]

Hello,

We have an Asterisk 11.3 server where we want log rotation handled purely
by Linux's logrotate, and not by Asterisk. To this end we've configured the
[general] action of /etc/asterisk/logger.conf with:

rotatestrategy = none

However, an "asterisk -rx 'logger reload'" still rotates the log files.
Does anyone know why?

Thank you in advance,

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[asterisk-users] VoIP support engineer opportunity

[David Cunningham]

Hello,

Voisonics is hiring a VoIP support engineer to assist our customers running
Asterisk based hosted PBX platforms. This is a part-time contract
work-from-home position.

For communication reasons we're looking for someone in a timezone
encompassing Far East Asia, Australia, New Zealand, Canada, the USA, and
Mexico. If you are not physically located in that area please do not apply
- being "flexible" from another part of the world is not what we're looking
for.

The role involves providing technical support of Asterisk based PBX
platforms to our customer's technical staff, Linux system administration,
and small dev-ops type development projects. It does not involve providing
technical support to end users or the general public.

Customers are located around the world. You will generally be responding
during your business hours, though sometimes out of hours work will be
necessary. Once training is completed, the position will involve providing
24x7 on-call emergency cover in rotation with other staff.

Must-have:
1. Fluent command-line Linux ability on Ubuntu, Debian, CentOS, and/or
RedHat.
2. Asterisk administration and configuration experience.
3. SIP debugging experience. For example, you should know what packets are
typically involved in setting up a call.
4. Experience with administration and configuration of Apache or MySQL.
5. Ability to program at least one language, such as Perl, Shell script, C,
Go, PHP, etc.
6. Good written and verbal English language ability.
7. Independant and critical thinking ability.
8. Have experience providing professional IT support to business.
9. Be an individual self-employed contractor.

Nice-to-have:
1. Experience with Kamailio, NFS, GlusterFS, Puppet, or Zabbix.
2. Javascript or AngularJS programming ability.
3. HTML and CSS programming ability.
4. Advanced network knowledge (beyond basic Linux networking which is a
must-have).

To apply for this role email me off-list. In your application:
1. List your experience/compatibility for each of the must-have
requirements individually, plus any of the nice-to-have items you fit as
well.
2. Provide your physical location, hours of availability, and indication of
hourly rate.
3. Let us know what other work you have during business hours.

Thank you,

-- 
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Re: [asterisk-users] Hangup hook to put back a call into a queue

[David P]

It might work for you to branch on ${DIALSTRING} just after your Dial
command, if you want to handle a BUSY, NOANSWER, or other result. But if
the peer of that Dial hungup, then based on what Joshua said, it seems
there's no recovery.
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[asterisk-users] Looking for sample hangup_handler_pop and _wipe using vars

[David P]

Please point me to samples of popping and wiping hangup handlers. I don't
need to use the values returned; I just need to clear any handlers before I
push a new one.

It's not clear at
https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers+Specification how
to provide vars on the right-hand side.

Cheers,
David
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[asterisk-users] Call disrupted...due to registration of third server?

[David P]

We use Asterisk 14 to proxy calls between two servers, 10.0.0.192 to
10.0.0.228. But sometimes another of our servers becomes listed as a SIP
agent, even though the server's IP address isn't part of our sip.conf,
extensions.conf, nor any other config I know of. For example in the log
snippet below, the source server experienced an SDP renegotiation in the
middle of a call, and seemingly as a consequence Asterisk re-locked on the
source and destination servers...but also registered third server
10.0.0.125. This seems to have broken the call to the desired destination
server.

[2020-01-14 18:08:25] VERBOSE[29350][C-0006] res_rtp_asterisk.c:
0x7f40240322e0 -- Strict RTP switching source address to 10.0.0.228:42150
[2020-01-14 18:08:26] VERBOSE[29324][C-0006] res_rtp_asterisk.c:
0x7f403c00c3b0 -- Strict RTP learning complete - Locking on source address
10.0.0.192:22522
[2020-01-14 18:08:26] VERBOSE[29350][C-0006] res_rtp_asterisk.c:
0x7f40240322e0 -- Strict RTP learning complete - Locking on source address
10.0.0.228:42150
[2020-01-14 18:09:01] VERBOSE[1389] asterisk.c: Remote UNIX connection
[2020-01-14 18:09:01] VERBOSE[29363] asterisk.c: Remote UNIX connection
disconnected
[2020-01-14 18:09:47] VERBOSE[1429] chan_sip.c: Registered SIP '1000' at
10.0.0.125:5060
[2020-01-14 18:09:47] VERBOSE[1429] chan_sip.c: Saved useragent
"FreeSWITCH-mod_sofia/1.6.20~64bit" for peer 1000

Is my description accurate for this log snippet?

How can we prevent the registration of third servers?
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Re: [asterisk-users] Handling a non-responsive peer after it answers

[David P]

Response below...

On Fri, Dec 27, 2019 at 12:02 PM David P  wrote:
>
> >
> > I'm looking for a way of detecting in my dialplan when a peer becomes
> > non-responsive after answering. [deleted] Is there a way to configure
> > a handler for this state?
> >
> > We use v14.7.6 and we dial the peer this way:
> >
> >  same =>
> >
> n,Set(CHANNEL(hangup_handler_push)=${CONTEXT},handleHangupByCaller,1(args))
> >  same =>
> >
> n,Dial(${AddressToReachPeer},2,b(${CONTEXT}^afterDialingPeerLogIpOfCb^1(${UUID}^${StartEpoch})))
> >  same => n,Goto(handle${DIALSTATUS},1)
> >
>

"Joshua C. Colp"  replied:

> [deleted] As for hanging up a call when the remote
> goes away that depends on the channel driver. For SIP both chan_sip and
> chan_pjsip provide session timers which use SIP messages to determine if
> the call is no longer valid, or RTP timeout which hangs up the call if
> media is not flowing for a period of time. These are configured in the
> respective channel driver configuration file.
>

Thanks, Joshua.

We want to check if a peer is responsive every few seconds, because it's a
person-to-bot call and we want to respond gracefully if the bot fails.

I tried adding
rtptimeout=4
to the config of the peer in sip.conf, but this causes hangup during the
person's turn.

Then I looked into session timers, and found that
https://issues.asterisk.org/jira/secure/attachment/28201/AsteriskSipSessionTimers.pdf
says the shortest period supported for such checks is 90 seconds, which is
much too long for us.

Is there another option? Would it allow calling a script or playing a
prompt on the way to hanging up?
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[asterisk-users] Handling a non-responsive peer after it answers

[David P]

I'm looking for a way of detecting in my dialplan when a peer becomes
non-responsive after answering. It seems that Asterisk knows when the peer
becomes non-responsive because it logs "Remote UNIX connection
disconnected" around the same time, and it seems that
if there is no following "Remote UNIX connection" within a short time, then
the peer can be considered non-responsive. Is there a way to configure a
handler for this state?

We use v14.7.6 and we dial the peer this way:

 same =>
n,Set(CHANNEL(hangup_handler_push)=${CONTEXT},handleHangupByCaller,1(args))
 same =>
n,Dial(${AddressToReachPeer},2,b(${CONTEXT}^afterDialingPeerLogIpOfCb^1(${UUID}^${StartEpoch})))
 same => n,Goto(handle${DIALSTATUS},1)

Cheers,
David
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[asterisk-users] One Touch Record and a matching entry in sip.conf

[David Cunningham]

Hello,

This has a bit of a long explanation... ultimately the question is why
adding a section to sip.conf made a difference to One Touch Record.

We're implementing a recording toggle using the "Record" button on a SIP
telephone and Asterisk's "One Touch Record" feature in features.conf.

It worked without problem when the calling party pressed the Record button,
but it didn't work when the called party pressed the Record button. When
the called party pressed the button Asterisk logged "Recording requested,
but no One Touch Monitor registered. (See features.conf)", even though it
was certainly enabled in features.conf

The destination was called with a dial string like SIP/@12.34.56.78:5060.
Eventually we worked out that adding a section to sip.conf like this
allowed it to start working:

[12.34.56.78:5060]
type = friend
... etc..

We already (before it started working) had a section almost identical but
without the port in the section name:
[12.34.56.78]
type = friend
... etc..

The question is, why did adding the section with the port in the name make
a difference? Is it because only Asterisk sip users (of which a friend is
one) are allowed to use One Touch Record? If so I don't see this documented
anywhere. Note that adding or removing the 'w' or 'W' options to the Dial
seems to make no difference at all, and following the addition of the
sip.conf section it works without either of these options in the Dial.

Thank you in advance for any insight into this.

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[asterisk-users] Music on hold depending on who put call on hold

[David Cunningham]

Hello,

Does anyone know of a way to play different music on hold depending on
which party puts the call on hold?

We can specify the music on hold per channel, but that doesn't do what is
needed. We want to play one music if the caller puts the call on hold, and
a different music if the called party puts the call on hold.

Thanks in advance for any assistance.

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Re: [asterisk-users] Find out which key ended recording?

[David Cunningham]

Hi Steve,

Thank you very much for that information. The result is the key in ascii
perfectly!


On Fri, 7 Jun 2019 at 18:05, Steve Edwards 
wrote:

> On Fri, 7 Jun 2019, David Cunningham wrote:
>
> > We're using Perl and so far I haven't found an equivalent there.
>
> On Thu, 6 Jun 2019, Steve Edwards wrote:
>
> > I'm not much of a Perl programmer...
>
> But you should never turn down an opportunity to develop your skills :)
>
> Try something like:
>
>  my $result = $AGI->record_file(
>'/tmp/foo'# filename
>  , 'wav' # format
>  , '#*0123456789'# escape digits
>  , '5000'# timeout
>  );
>  $AGI->verbose('result =  ' . $result, 0);
>
> Which results in:
>
> AGI Rx << RECORD FILE /tmp/foo wav #*0123456789
> 5000
> AGI Tx >> 200 result=50 (dtmf) endpos=0
> AGI Rx << VERBOSE "result =  50"
>
> when '2' is pressed.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>  https://www.linkedin.com/in/steve-edwards-4244281
>
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] Find out which key ended recording?

[David Cunningham]

Hi Steve,

What language is that please? We're using Perl and so far I haven't found
an equivalent there.

Thanks for your help.


On Fri, 7 Jun 2019 at 12:10, Steve Edwards 
wrote:

> On Fri, 7 Jun 2019, David Cunningham wrote:
>
> > We have a need to record audio and allow the user to press any DTMF key
> > to end the recording. Currently we're using the AGI command "record
> > file" which does allow us to specify which DTMF keys can end the
> > recording.
> >
> > However we also need to know which key actually ended the recording.
> > Note that only allowing # or * to end the recording won't work for us.
> >
> > Does anyone know how we can tell which key ended the recording? Thanks
> > in advance for any help.
>
> Here's a snippet from one of my AGIs:
>
> // record the voice
>  exec_agi("RECORD FILE"
>" %s" // filename
>" wav"// format
>" #*1234567890"
>  // escape digits
>" %d000"  // timeout in ms
>" BEEP"   // BEEP
>  , recorded_path
>  , recording_limit
>  );
>
> // should we abort?
>  if  ('*' == agi_environment.result)
>  {
>  agi_set_variable("STATUS", "*");
>  exit(EXIT_SUCCESS);
>  }
>
> // are we finished?
>  if  ('#' == agi_environment.result)
>  {
>  break;
>  }
>
> Looks like agi_environment.result is your Huckleberry.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>  https://www.linkedin.com/in/steve-edwards-4244281
>
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[asterisk-users] Find out which key ended recording?

[David Cunningham]

Hello,

We have a need to record audio and allow the user to press any DTMF key to
end the recording. Currently we're using the AGI command "record file"
which does allow us to specify which DTMF keys can end the recording.

However we also need to know *which* key actually ended the recording. Note
that only allowing # or * to end the recording won't work for us.

Does anyone know how we can tell which key ended the recording? Thanks in
advance for any help.

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
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Re: [asterisk-users] pjsip and tls client: How to decrypt Wireshark trace?

[David Cunningham]

Hi Michael,

If you can get a copy of the private key you can import that to Wireshark
and see the encrypted information:
https://support.citrix.com/article/CTX116557


On Sun, 12 May 2019 at 04:55, Michael Maier  wrote:

> Hello!
>
> I'm just wondering if it's possible to decrypt sips packages in Wireshark
> while asterisk runs as sips client (connecting to the provider w/
> tls 1.2)? I don't use an own certificate.
>
>
> Thanks
> Michael
>
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[asterisk-users] Change of H264 profile level problem

[David Cunningham]

Hello,

Can anyone help with an issue regarding the H264 profile level being passed
through Asterisk? We have a video call like this:

Caller A -> Asterisk -> Called B

Caller A's INVITE SDP offers "profile-level-id=42801f", and Called B
replies a 200 OK containing "profile-level-id=42801e" in its SDP.  Note
that this ends with an 'e' rather than an 'f'. The problem is that Asterisk
forwards the 200 OK to Caller A with "profile-level-id=42801f" in the SDP,
so not what Called B sent. Caller A then starts transmitting video in a
resolution that Called B can't handle, and the video is displayed as blank.

My question is, since Asterisk doesn't do video transcoding, why doesn't it
pass though Called B's "profile-level-id=42801e" unchanged? If it did then
Caller A might use a resolution that Called B can handle. We are using
Asterisk 11.25.3.

Thanks in advance for any assistance.

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[asterisk-users] Reliable information on which SIP party is transferring call

[David Cunningham]

Hello,

Can anyone advise whether there's some method to reliably determine which
SIP device is the party performing a transfer (REFER or using
features.conf), and to whom they're transferring?

We've been analysing the AMI AttendedTransfer events and of course an
extension isn't necessarily a SIP device, and while we can usually figure
one from the other for one transfer, for two or more transfers it appears
to be a mess of information.

Is there some secret to figuring out a clean set of rules? Ideally for our
purpose the events would closely resemble those in the SIP packets
themselves. Thanks in advance for any help.

-- 
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[asterisk-users] ChanSpy "Audiohook has stale audio in its factories" problem

[David Cunningham]

Hello,

We have an issue where using ChanSpy with the 'B' option to talk to both
parties doesn't work in some call scenarios (calling out to a T-mobile
device). The problem is that the spying party can talk to the called person
but not the caller. The Asterisk log in this case records warnings as
below, which then continue for the life of the spying call.

Would anyone know the cause of this? We are using Asterisk 11.25.3. Thanks
in advance for any advice.

[Jan 18 15:59:39] DEBUG[39503][C-0bbd] autochan.c: Created autochan
0x152564004ea0 to hold channel SIP/15-xx.yy.122.136-0c85
(0x15251806fb28)
[Jan 18 15:59:39] NOTICE[39503][C-0bbd] app_chanspy.c: Attaching
SIP/enswitch-local-0c86 to SIP/15-xx.yy.122.136-0c85
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Audiohook
0x1525f5d59c58 has stale audio in its factories. Flushing them both
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Audiohook
0x1525f5d5b6b0 has stale audio in its factories. Flushing them both
[Jan 18 15:59:39] DEBUG[38605][C-0bbc] audiohook.c: Audiohook
0x1525f5d58200 has stale audio in its factories. Flushing them both
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160
samples from write factory 0x1525f5d58ec8
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] res_rtp_asterisk.c: Difference
is 888, ms is 131
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160
samples from write factory 0x1525f5d58ec8
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Read factory
0x1525f5d58288 and write factory 0x1525f5d58ec8 both fail to provide 160
samples
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160
samples from write factory 0x1525f5d58ec8
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160
samples from write factory 0x1525f5d58ec8
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Audiohook
0x1525f5d59c58 has stale audio in its factories. Flushing them both
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160
samples from write factory 0x1525f5d58ec8
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160
samples from write factory 0x1525f5d58ec8
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160
samples from write factory 0x1525f5d58ec8
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160
samples from write factory 0x1525f5d58ec8
[Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160
samples from write factory 0x1525f5d58ec8

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
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Re: [asterisk-users] CURL to post application/json

[David P]

Thanks, Nasir, I'll see if that allows us to avoid SHELL.

On Fri, 5 Oct 2018, 4:53 pm Nasir Iqbal,  wrote:

> Hi David,
>
> Have you tried CURLOPT function.
> i.e
> Set(CURLOPT(header)=Content-Type: application/json)
>
> Regards
>
> Nasir Iqbal
>
> ICTBroadcast - an Auto Dialer software for ITSP
> <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition>
> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
> http://www.ictbroadcast.com/
>
>
> On Fri, Oct 5, 2018 at 1:59 AM David P  wrote:
>
>> We tried to use the CURL fn to POST json, but it's sent as form data and
>> there seems no support for changing the Content-Type header. We switched to
>> invoking curl in the shell.
>>
>> All the documentation I could find says there is just one parameter for
>> the url and an optional second for POST body. Is there an undocumented way
>> to set Content-Type?
>> --
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>> Astricon is coming up October 9-11!  Signup is available at:
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>>
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>>
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[asterisk-users] CURL to post application/json

[David P]

We tried to use the CURL fn to POST json, but it's sent as form data and
there seems no support for changing the Content-Type header. We switched to
invoking curl in the shell.

All the documentation I could find says there is just one parameter for the
url and an optional second for POST body. Is there an undocumented way to
set Content-Type?
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Re: [asterisk-users] WebRTC as Softphone substitute ?

[David P]

Thanks for sharing this, Alex. It sounds like TURN, as a media repeater,
wouldn't work if the media must be secured (via SRTP). Is that right?

On Wed, 3 Oct 2018, 3:17 pm alex epshteyn,  wrote:

> WebRTC requires a few specific things to be in place. We have blog posts
> that talk about WebRTC based Thirdlane Connect, but most of the information
> applies to WebRTC applications in general.
>
> https://www.thirdlane.com/blog/get-up-and-running-with-thirdlane-connect
>
> https://www.thirdlane.com/blog/nat-stun-turn-and-ice
>
> Best regards,
> Alex
>
>
> Alex Epshteyn
> a...@thirdlane.com
> +1 (415) 261 6601
> www.thirdlane.com
>
>
> On Oct 2, 2018, at 6:08 PM, Nasir Iqbal  wrote:
>
> @Olivior
> I agree that seting up WebRTC is hard, however when done it is smooth to
> use. For replication you can build RPMs with working configurations.
>
> Regarding stability, it is not being used widly, so can't say it is
> mature. However we have no complain so far regarding audio or connectivity.
> sometime we provide support for "allow media / mic" type issues, but you
> know it is security feature and not a bug.
>
> Regards
>
> On Tue, Oct 2, 2018, 13:03 Olivier  wrote:
>
>> @Nasir:
>> Thanks for replying here.
>>
>> Did you met in your deployments, the kind of stability issues Carlos
>> reported earlier ?
>>
>> Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal  a
>> écrit :
>>
>>> Hi Olivior,
>>>
>>> We have recently worked on a WebRTC based agent panel. As based on my
>>> experience I think that WebRTC based phones are far better and cheaper then
>>> those soft / sip phone. the big plus is that they are easy to customize and
>>> developer can use the power of browser and web to build / offer features
>>> which are not possible with regular phones.
>>>
>>> Regarding your concern about BLF or call history, for me as a being
>>> developer it is just a matter of customization.
>>>
>>> Regards
>>>
>>> Nasir Iqbal
>>>
>>> ICTBroadcast - an Auto Dialer software for ITSP
>>> 
>>> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
>>> http://www.ictbroadcast.com/
>>>
>>>
>>> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez 
>>> wrote:
>>>
 On 9/26/18 10:20 AM, Matthew Fredrickson wrote:

 > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez 
 wrote:
 >> On 9/26/2018 4:46 AM, Olivier wrote:
 >>
 >>> Hello,
 >>>
 >>> This morning, I asked myself if WebRTC could be a viable alternative
 >>> to softphone deployment.
 >>>
 >>> For me, main issue with Softphones is the amount of work needed for
 >>> installation and configuration.
 >>> Also, Softphones must be carefully choosen if Deskphone-like quality
 >>> is expected.
 >>>
 >>> Now that WebRTC becomes ubiquitous, it might make sense to trade
 >>> Softphone features (call history, BLF, ...) for WebRTC deployment
 >>> simplicity.
 >>>
 >>> What do you think of this ?
 >>> What kind of experience did you met with such WebRTC deployments ?
 >>> What about classic telephony features (CallTransfer) ?
 >>> Have you tried Cyber Maga Phone 2K ?
 >>>
 >>   If you can get it to work WebRTC is a good option.  The
 problem is
 >> that any changes in your network may disrupt it and even trying to
 >> replicate your installation is difficult.  I have it working fine on
 my
 >> website so customers can call us directly from our web page but I
 never
 >> could get Cyber Mega Phone 2K to work on the same server.  We used
 JSSIP
 >> to create the webrtc phone on our website.
 > We just updated the documentation for how to get CMP2K working on the
 > wiki [1].  We'd love some feedback if you still have issues getting it
 > setup so that we can improve the docs.
 >
 > [1]
 https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone
 >
 > Best wishes,
 > Matthew Fredrickson
 >
  I followed the procedure indicated in the link but I cannot get
 remote video.  I can only see my own feed.  We do have audio for a
 little while.  For some reason the users get disconnected after a few
 minutes even though you can still see your video feed on screen.  This
 was done with Asterisk 15.6.0

 --
 Telecomunicaciones Abiertas de México S.A. de C.V.
 Carlos Chávez
 +52 (55)8116-9161


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Re: [asterisk-users] How to know the IP of "manager show connected" in dialplan

[David Duffett]

How about using the CUT() function to get the IP address from the return
from running the System() application running asterisk -rx "manager show
connected"?

I'm not in front of a machine, so cannot test this out...



On Wed, 25 Jul 2018, 15:42 Ludovic Gasc,  wrote:

> Maybe I'm wrong, but, with the information you give us, for me, it seems
> more elegant to use FastAGI to be sure to communicate with the right remote
> process.
>
> As Antony suggested, UserEvent is also an option, except if you have
> several dialers connected at the same time or if you need to have an
> acknowledge that the action is correctly launched.
>
> Regards.
> --
> Ludovic Gasc (GMLudo)
>
>
> Le mer. 25 juil. 2018 à 19:54, Saint Michael  a écrit :
>
>> ​I need to launch a remote process at the machine that has the dialer. I
>> could
>> hard-code the IP address in a global variable, but It would be much more
>> elegant if the dialplan would have a "manager" object where I could read
>> "manager-->connected". ​
>>
>>
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>
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Re: [asterisk-users] How to steal an answered call?

[David Cunningham]

Hello Patrick and others,

Thanks, I wasn't familiar with the Bridge application and it may allow us
to do what's needed.

A transfer would of course be simpler but the user wants what the user
wants...

Thank you.


On 9 July 2018 at 19:52, John Kiniston  wrote:

> David,
>
> You should be able to use the Bridge dialplan application to do what you
> want.
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Bridge
>
> I use the CHANNELS function and the IMPORT function to find the channel to
> bridge to my caller.
>
>
> On Sun, Jul 8, 2018 at 8:17 PM David Cunningham 
> wrote:
>
>> Hello,
>>
>> I'm familiar with Pickup/PickupChan for taking a ringing call, but does
>> anyone know how a phone can "steal" an already answered call from another
>> phone? Our users have decided that call parking is too long-winded and
>> don't want to use that.
>>
>> For example: phone A calls phone B, phone B answers the call, phone C
>> dials something to "steal" the call from B, and finally A and C are talking.
>>
>> Searching on voip-info.org shows a "BristuffSteal" command but it's very
>> out of date (Asterisk 1.2).
>>
>> Thanks in advance for any suggestions.
>>
>> Kind regards,
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>> --
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>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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>> To UNSUBSCRIBE or update options visit:
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>
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
>
> --
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> org/
>
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>
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[asterisk-users] How to steal an answered call?

[David Cunningham]

Hello,

I'm familiar with Pickup/PickupChan for taking a ringing call, but does
anyone know how a phone can "steal" an already answered call from another
phone? Our users have decided that call parking is too long-winded and
don't want to use that.

For example: phone A calls phone B, phone B answers the call, phone C dials
something to "steal" the call from B, and finally A and C are talking.

Searching on voip-info.org shows a "BristuffSteal" command but it's very
out of date (Asterisk 1.2).

Thanks in advance for any suggestions.

Kind regards,

--
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http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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Re: [asterisk-users] Only 8kHz recorded after disallowing all but G722 codec on inbound

[David P]

To followup my question below, we're looking for a way to record in
Asterisk at 16kHz but send only 8kHz (ulaw) to our peers. Is this possible?

Btw, I wanted to ask this as a followup to
https://community.asterisk.org/t/change-sample-rate-to-16khz/73842/2 but
whenever I try to login to that site, I get "Sorry, there was an error
authorizing your account. Perhaps you did not approve authorization?" I've
never received an email asking to verify my address, if that's what this
error means. I just tried re-registering, too.

On Sun, Jun 17, 2018 at 7:25 PM David P  wrote:

> I also just tried adding this:
>
>  same => n,Set(SIP_CODEC_INBOUND=g722)
>
> On Sat, Jun 16, 2018 at 4:36 PM David P  wrote:
>
>> We want to record inbound channels at 16kHz, but send only 8kHz to our
>> peers. I've set our default profile in sip.conf to disallow all but g722,
>> and the peers disallow all but ulaw. We have a proxy in front of Asterisk
>> that is configured to disallow all but G722 also.
>>
>> My test calls show inbound to the proxy is recorded at 16kHz, inbound in
>> Asterisk is only 8kHz, and the peers receive 8kHz. So the only thing not
>> working is Asterisk's sampling rate on inbound, and it seems to be
>> downsampling.
>>
>> After a lot of web searching, I can't find any explanation of why we're
>> not getting 16kHz for G722. We're using Asterisk 14.7.6.
>>
>> Cheers,
>> David
>>
>
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Re: [asterisk-users] Only 8kHz recorded after disallowing all but G722 codec on inbound

[David P]

I also just tried adding this:

 same => n,Set(SIP_CODEC_INBOUND=g722)

On Sat, Jun 16, 2018 at 4:36 PM David P  wrote:

> We want to record inbound channels at 16kHz, but send only 8kHz to our
> peers. I've set our default profile in sip.conf to disallow all but g722,
> and the peers disallow all but ulaw. We have a proxy in front of Asterisk
> that is configured to disallow all but G722 also.
>
> My test calls show inbound to the proxy is recorded at 16kHz, inbound in
> Asterisk is only 8kHz, and the peers receive 8kHz. So the only thing not
> working is Asterisk's sampling rate on inbound, and it seems to be
> downsampling.
>
> After a lot of web searching, I can't find any explanation of why we're
> not getting 16kHz for G722. We're using Asterisk 14.7.6.
>
> Cheers,
> David
>
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[asterisk-users] Only 8kHz recorded after disallowing all but G722 codec on inbound

[David P]

We want to record inbound channels at 16kHz, but send only 8kHz to our
peers. I've set our default profile in sip.conf to disallow all but g722,
and the peers disallow all but ulaw. We have a proxy in front of Asterisk
that is configured to disallow all but G722 also.

My test calls show inbound to the proxy is recorded at 16kHz, inbound in
Asterisk is only 8kHz, and the peers receive 8kHz. So the only thing not
working is Asterisk's sampling rate on inbound, and it seems to be
downsampling.

After a lot of web searching, I can't find any explanation of why we're not
getting 16kHz for G722. We're using Asterisk 14.7.6.

Cheers,
David
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Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

[David P]

FYI, we found that our peers don't hangup properly. But we would still like
to know how to get the peer's hangup handler to fire upon peer hangup,
because right now it corrupts our globals by firing after the caller's
hangup handler.

On Tue, Jun 5, 2018 at 5:40 PM, David P  wrote:

> FWIW, I added the following after the Dial, and it doesn't appear in cli
> after peer hangup:
>
> same => n,NoOp(After Dial ${AddressToReachPeer})
>
> I also tried putting 'g' before the 'b'.
>
> I also tried removing the context headers of the hangup handlers and
> predial handler, and just referring to those by extensions. No difference.
>
> On Tue, Jun 5, 2018 at 3:17 PM, David P  wrote:
>
>> This has been super-helpful, Eric. However, the handleHangupByPeer priorities
>> below are still not run when the peer hangs-up. The last line in the cli
>> when the peer hangs-up is still:
>> Strict RTP learning complete - Locking on source address
>> (Although sometimes there is also: Retransmission timeout reached on
>> transmission)
>>
>>  same => 
>> n(callPeer),Set(GLOBAL(CB${IndexIntoPeers}CurrentCallsCount)=$[${PeerCurrentCallsCount}
>> + 1])
>>  ; Ensure that hangup by caller/inbound-channel will invoke
>> handleHangupByCaller.
>>  same => n,Set(CHANNEL(hangup_handler_push)=handleHangupByCaller,s,1(
>> args))
>>  same => n,Set(AddressToReachPeer=SIP/${EXTEN:0:4}@${PeerBeingConside
>> red})
>>  ; Ensure that the channel of the peer (i.e. outbound-channel) is
>> configured with hangup handler.
>>  same => n,Dial(${AddressToReachPeer},,b(beforeDialingPeerConfigureIt
>> sChannelForPeerHangupHandling^s^1))
>>  same => n,Hangup
>>
>> [beforeDialingPeerConfigureItsChannelForPeerHangupHandling]
>> exten => s,1,Set(CHANNEL(hangup_handler_push)=handleHangupByPeer,s,1(
>> args))
>>  same => n,Return
>>
>> [handleHangupByPeer]
>>  ; Ensure that GLOBAL(CB${IndexIntoPeers}CurrentCallsCount) is
>> decremented after hangup, and end-of-call-epoch is set.
>> exten => s,1,NoOp(${PeerBeingConsidered} peer channel: Entered
>> handleHangupByCallerOrPeer Calls ${CB${IndexIntoPeers}CurrentCallsCount})
>>  same => n,Set(GLOBAL(CB${IndexIntoPeers}CurrentCallsCount)=$[${CB${I
>> ndexIntoPeers}CurrentCallsCount} - 1])
>>  same => n,Set(GLOBAL(${PeerBeingConsidered}EpochAtCallEnd)=${EPOCH})
>>  same => n,Return
>>
>> [handleHangupByCaller]
>>  ; Ensure that GLOBAL(CB${IndexIntoPeers}CurrentCallsCount) is
>> decremented after hangup, and end-of-call-epoch is set.
>> exten => s,1,NoOp(${PeerBeingConsidered} caller channel: Entered
>> handleHangupByCallerOrPeer Calls ${CB${IndexIntoPeers}CurrentCallsCount})
>>  same => n,Set(GLOBAL(CB${IndexIntoPeers}CurrentCallsCount)=$[${CB${I
>> ndexIntoPeers}CurrentCallsCount} - 1])
>>  same => n,Set(GLOBAL(${PeerBeingConsidered}EpochAtCallEnd)=${EPOCH})
>>  same => n,Return
>>
>>
>> When the caller hangs-up, handleHangupByCaller is run first, then 
>> handleHangupByPeer
>> runs. (And strangely, the value of global 
>> CB${IndexIntoPeers}CurrentCallsCount
>> isn't accessible in handleHangupByPeer.)
>>
>> Cheers,
>> David
>>
>>
>> On Tue, Jun 5, 2018 at 12:58 PM, Eric Wieling  wrote:
>>
>>> Don't use the _. pattern.  Ever.
>>>
>>> The call has two channels so it needs two hangup handlers, something
>>> like this, though I've not tested it.
>>>
>>> [some_context]
>>> exten => _X.,1,Noop
>>>  same => n,Set(CHANNEL(hangup_handler_push)=my_caller_hangup_handler)
>>>  same => n,Dial(SIP/number@peer,b(pre_dial^s^1))
>>>  same => n,Hangup
>>>
>>> [pre_dial]
>>> exten => s,1,Set(CHANNEL(hangup_handler_push)=my_called_hangup_handler)
>>>  same => Return
>>>
>>> See: https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
>>> and https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers
>>>
>>>
>
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Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

[David P]

FWIW, I added the following after the Dial, and it doesn't appear in cli
after peer hangup:

same => n,NoOp(After Dial ${AddressToReachPeer})

I also tried putting 'g' before the 'b'.

I also tried removing the context headers of the hangup handlers and
predial handler, and just referring to those by extensions. No difference.

On Tue, Jun 5, 2018 at 3:17 PM, David P  wrote:

> This has been super-helpful, Eric. However, the handleHangupByPeer priorities
> below are still not run when the peer hangs-up. The last line in the cli
> when the peer hangs-up is still:
> Strict RTP learning complete - Locking on source address
> (Although sometimes there is also: Retransmission timeout reached on
> transmission)
>
>  same => 
> n(callPeer),Set(GLOBAL(CB${IndexIntoPeers}CurrentCallsCount)=$[${PeerCurrentCallsCount}
> + 1])
>  ; Ensure that hangup by caller/inbound-channel will invoke
> handleHangupByCaller.
>  same => n,Set(CHANNEL(hangup_handler_push)=handleHangupByCaller,s,
> 1(args))
>  same => n,Set(AddressToReachPeer=SIP/${EXTEN:0:4}@${PeerBeingConsidered})
>  ; Ensure that the channel of the peer (i.e. outbound-channel) is
> configured with hangup handler.
>  same => n,Dial(${AddressToReachPeer},,b(beforeDialingPeerConfigureItsC
> hannelForPeerHangupHandling^s^1))
>  same => n,Hangup
>
> [beforeDialingPeerConfigureItsChannelForPeerHangupHandling]
> exten => s,1,Set(CHANNEL(hangup_handler_push)=
> handleHangupByPeer,s,1(args))
>  same => n,Return
>
> [handleHangupByPeer]
>  ; Ensure that GLOBAL(CB${IndexIntoPeers}CurrentCallsCount) is
> decremented after hangup, and end-of-call-epoch is set.
> exten => s,1,NoOp(${PeerBeingConsidered} peer channel: Entered
> handleHangupByCallerOrPeer Calls ${CB${IndexIntoPeers}CurrentCallsCount})
>  same => n,Set(GLOBAL(CB${IndexIntoPeers}CurrentCallsCount)=$[${CB${
> IndexIntoPeers}CurrentCallsCount} - 1])
>  same => n,Set(GLOBAL(${PeerBeingConsidered}EpochAtCallEnd)=${EPOCH})
>  same => n,Return
>
> [handleHangupByCaller]
>  ; Ensure that GLOBAL(CB${IndexIntoPeers}CurrentCallsCount) is
> decremented after hangup, and end-of-call-epoch is set.
> exten => s,1,NoOp(${PeerBeingConsidered} caller channel: Entered
> handleHangupByCallerOrPeer Calls ${CB${IndexIntoPeers}CurrentCallsCount})
>  same => n,Set(GLOBAL(CB${IndexIntoPeers}CurrentCallsCount)=$[${CB${
> IndexIntoPeers}CurrentCallsCount} - 1])
>  same => n,Set(GLOBAL(${PeerBeingConsidered}EpochAtCallEnd)=${EPOCH})
>  same => n,Return
>
>
> When the caller hangs-up, handleHangupByCaller is run first, then 
> handleHangupByPeer
> runs. (And strangely, the value of global CB${IndexIntoPeers}CurrentCallsCount
> isn't accessible in handleHangupByPeer.)
>
> Cheers,
> David
>
>
> On Tue, Jun 5, 2018 at 12:58 PM, Eric Wieling  wrote:
>
>> Don't use the _. pattern.  Ever.
>>
>> The call has two channels so it needs two hangup handlers, something like
>> this, though I've not tested it.
>>
>> [some_context]
>> exten => _X.,1,Noop
>>  same => n,Set(CHANNEL(hangup_handler_push)=my_caller_hangup_handler)
>>  same => n,Dial(SIP/number@peer,b(pre_dial^s^1))
>>  same => n,Hangup
>>
>> [pre_dial]
>> exten => s,1,Set(CHANNEL(hangup_handler_push)=my_called_hangup_handler)
>>  same => Return
>>
>> See: https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
>> and https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers
>>
>>
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Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

[David P]

This has been super-helpful, Eric. However, the handleHangupByPeer priorities
below are still not run when the peer hangs-up. The last line in the cli
when the peer hangs-up is still:
Strict RTP learning complete - Locking on source address
(Although sometimes there is also: Retransmission timeout reached on
transmission)

 same =>
n(callPeer),Set(GLOBAL(CB${IndexIntoPeers}CurrentCallsCount)=$[${PeerCurrentCallsCount}
+ 1])
 ; Ensure that hangup by caller/inbound-channel will invoke
handleHangupByCaller.
 same => n,Set(CHANNEL(hangup_handler_push)=handleHangupByCaller,s,1(args))
 same => n,Set(AddressToReachPeer=SIP/${EXTEN:0:4}@${PeerBeingConsidered})
 ; Ensure that the channel of the peer (i.e. outbound-channel) is
configured with hangup handler.
 same =>
n,Dial(${AddressToReachPeer},,b(beforeDialingPeerConfigureItsChannelForPeerHangupHandling^s^1))
 same => n,Hangup

[beforeDialingPeerConfigureItsChannelForPeerHangupHandling]
exten => s,1,Set(CHANNEL(hangup_handler_push)=handleHangupByPeer,s,1(args))
 same => n,Return

[handleHangupByPeer]
 ; Ensure that GLOBAL(CB${IndexIntoPeers}CurrentCallsCount) is decremented
after hangup, and end-of-call-epoch is set.
exten => s,1,NoOp(${PeerBeingConsidered} peer channel: Entered
handleHangupByCallerOrPeer Calls ${CB${IndexIntoPeers}CurrentCallsCount})
 same =>
n,Set(GLOBAL(CB${IndexIntoPeers}CurrentCallsCount)=$[${CB${IndexIntoPeers}CurrentCallsCount}
- 1])
 same => n,Set(GLOBAL(${PeerBeingConsidered}EpochAtCallEnd)=${EPOCH})
 same => n,Return

[handleHangupByCaller]
 ; Ensure that GLOBAL(CB${IndexIntoPeers}CurrentCallsCount) is decremented
after hangup, and end-of-call-epoch is set.
exten => s,1,NoOp(${PeerBeingConsidered} caller channel: Entered
handleHangupByCallerOrPeer Calls ${CB${IndexIntoPeers}CurrentCallsCount})
 same =>
n,Set(GLOBAL(CB${IndexIntoPeers}CurrentCallsCount)=$[${CB${IndexIntoPeers}CurrentCallsCount}
- 1])
 same => n,Set(GLOBAL(${PeerBeingConsidered}EpochAtCallEnd)=${EPOCH})
 same => n,Return


When the caller hangs-up, handleHangupByCaller is run first, then
handleHangupByPeer
runs. (And strangely, the value of global CB${IndexIntoPeers}CurrentCallsCount
isn't accessible in handleHangupByPeer.)

Cheers,
David


On Tue, Jun 5, 2018 at 12:58 PM, Eric Wieling  wrote:

> Don't use the _. pattern.  Ever.
>
> The call has two channels so it needs two hangup handlers, something like
> this, though I've not tested it.
>
> [some_context]
> exten => _X.,1,Noop
>  same => n,Set(CHANNEL(hangup_handler_push)=my_caller_hangup_handler)
>  same => n,Dial(SIP/number@peer,b(pre_dial^s^1))
>  same => n,Hangup
>
> [pre_dial]
> exten => s,1,Set(CHANNEL(hangup_handler_push)=my_called_hangup_handler)
>  same => Return
>
> See: https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
> and https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers
>
>
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Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

[David P]

Thanks, Eric. I just tried a hangup handler, but it's showing a similar
problem: When the peer hangs-up, the hangup handler is not invoked and the
caller channel remains open.

 same =>
n(callPeer),Set(GLOBAL(Peer${IndexIntoPeers}CurrentCallsCount)=$[${PeerCurrentCallsCount}
+ 1])
 same =>
n,Set(CHANNEL(hangup_handler_push)=handleHangupByCallerOrPeer,doesntMatter,1(args))
 same => n,Set(DialForPeer=SIP/${EXTEN:0:4}@${PeerBeingConsidered})
 same => n,Dial(${DialForPeer})
 same => n,Hangup()

[handleHangupByCallerOrPeer]
exten => _.,1,NoOp(${PeerBeingConsidered}: Entered
handleHangupByCallerOrPeer Calls ${Peer${IndexIntoPeers}CurrentCallsCount})
 same =>
n,Set(GLOBAL(Peer${IndexIntoPeers}CurrentCallsCount)=$[${Peer${IndexIntoPeers}CurrentCallsCount}
- 1])
 same => n,Set(GLOBAL(${PeerBeingConsidered}EpochAtCallEnd)=${EPOCH})
 same => n,Return()

I've also tried replacing the Dial above with:

 same => n,Dial(${DialForPeer},,g)

Cheers,
David

On Tue, Jun 5, 2018 at 7:38 AM, Eric Wieling  wrote:

> Use hangup handlers, they work around the issues with the 'h' extension.
>
> On 06/05/2018 05:33 AM, David P wrote:
>
>> Thanks, Anthony.
>>
>> I added both 'g' and 'F' options. Now, when the caller hangs-up, my
>> cleanup code is run by both the caller channel and the peer channel, but I
>> only want the caller channel to do that.
>>
>> Also, when the peer hangs-up, there is no execution of the priorities
>> following the Dial.
>>
>> Finally, is there a way to reset all globals, maybe as a variant of
>> "dialplan reload"?
>>
>> On Tue, Jun 5, 2018 at 1:21 AM, Antony Stone <
>> antony.st...@asterisk.open.source.it <mailto:Antony.Stone@asterisk.
>> open.source.it>> wrote:
>>
>> On Tuesday 05 June 2018 at 08:33:26, David P wrote:
>>
>> > We're using Asterisk 14.7.6 and I have a dialplan that ends like
>> this:
>> > >  same => n,Dial(SIP/${EXTEN:0:4}@peer1)
>> >  same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH})
>> >  same => n,Hangup()
>> > > When peer1 hangsup, the priorities after the Dial are
>> executed fine. But
>> > when the caller hangsup during the Dial, the cleanup steps aren't
>> done.
>> > Why?
>> > > I did read "Note that on a successful connection, in the
>> absence of the g
>> > and G modifiers (below), the Dial command does not return to allow
>> > execution of further commands for that extension in that context."
>> at
>> > https://www.voip-info.org/asterisk-cmd-dial/
>> <https://www.voip-info.org/asterisk-cmd-dial/> But it seems not to
>> apply
>> > because I'm seeing the 'g' behavior without specifying that option,
>> and the
>> > 'G' option seems intended for a far more complicated scenario.
>>
>> If you're getting "g" functionality without specifying it,
>> congratulations.
>>
>> If you want something similar when the callER hangs up, you want to
>> use the F
>> option.
>>
>> Regards,
>>
>>
>> Antony.
>>
>>
>>
>>
> --
> http://help.nyigc.net/
>
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[asterisk-users] remove

[David Mutterer]

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Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

[David P]

Thanks, Anthony.

I added both 'g' and 'F' options. Now, when the caller hangs-up, my cleanup
code is run by both the caller channel and the peer channel, but I only
want the caller channel to do that.

Also, when the peer hangs-up, there is no execution of the priorities
following the Dial.

Finally, is there a way to reset all globals, maybe as a variant of
"dialplan reload"?

On Tue, Jun 5, 2018 at 1:21 AM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Tuesday 05 June 2018 at 08:33:26, David P wrote:
>
> > We're using Asterisk 14.7.6 and I have a dialplan that ends like this:
> >
> >  same => n,Dial(SIP/${EXTEN:0:4}@peer1)
> >  same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH})
> >  same => n,Hangup()
> >
> > When peer1 hangsup, the priorities after the Dial are executed fine. But
> > when the caller hangsup during the Dial, the cleanup steps aren't done.
> > Why?
> >
> > I did read "Note that on a successful connection, in the absence of the g
> > and G modifiers (below), the Dial command does not return to allow
> > execution of further commands for that extension in that context." at
> > https://www.voip-info.org/asterisk-cmd-dial/ But it seems not to apply
> > because I'm seeing the 'g' behavior without specifying that option, and
> the
> > 'G' option seems intended for a far more complicated scenario.
>
> If you're getting "g" functionality without specifying it, congratulations.
>
> If you want something similar when the callER hangs up, you want to use
> the F
> option.
>
> Regards,
>
>
> Antony.
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[asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

[David P]

We're using Asterisk 14.7.6 and I have a dialplan that ends like this:

 same => n,Dial(SIP/${EXTEN:0:4}@peer1)
 same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH})
 same => n,Hangup()

When peer1 hangsup, the priorities after the Dial are executed fine. But
when the caller hangsup during the Dial, the cleanup steps aren't done. Why?

I did read "Note that on a successful connection, in the absence of the g
and G modifiers (below), the Dial command does not return to allow
execution of further commands for that extension in that context." at
https://www.voip-info.org/asterisk-cmd-dial/ But it seems not to apply
because I'm seeing the 'g' behavior without specifying that option, and the
'G' option seems intended for a far more complicated scenario.

Cheers,
David
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[asterisk-users] Queue of automated members

[David P]

We're using Asterisk 14.7.6, and we're able to route particular extensions
to a corresponding automated member via SIP. Each extension has its own
specific content, and all the automated members are configured to handle
any of these extensions. We would like to achieve some scaling by putting
all the members in a single queue.

Here are some problems we've run into:

1) How should these members be declared in queues.conf? We couldn't find
much about automated members in the Asterisk Book (
http://the-asterisk-book.com/1.6/queues.conf.html ) nor the wiki (
https://wiki.asterisk.org/wiki/display/AST/Building+Queues).

For example, all of our extensions start with 773, and we're starting with
two automated members (declared in sip.conf as CB1 and CB2).

member => SIP/773X@CB1
member => SIP/773X@CB2

Is this a valid way to declare these members and the extensions they can
handle?

2) When I configure the queue that way, it looks like calls don't make it
to the automated agents. Is there a way to inspect Asterisk to see what
extension is being provided to the agents by the queue?

  == Using SIP RTP CoS mark 5
   > 0x7f32f8013380 -- Strict RTP learning after remote address set to:
(IP address of caller):24804
-- Executing [77314a31e9c0-6173-4b41-9938-be3325139088@default:1]
Answer("SIP/1000-0012", "") in new stack
   > 0x7f32f8013380 -- Strict RTP switching to RTP target address (IP
address of caller):24804 as source
-- Auto fallthrough, channel 'SIP/1000-0012' status is 'UNKNOWN'

Cheers,
David
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[asterisk-users] Long extensions that contain dashes

[David P]

We would like to use 20-char extension values that use dashes and alphanums
after the first four digits. In order to handle these via pattern-matching,
how can I define a pattern that allows dashes? There seems to be no option
at http://the-asterisk-book.com/1.6/einleitung-regex.html#re
gular-expression-syntax However, when I try a period, it seems to match the
long suffix including the dashes. I want to know whether to depend on this
continuing to work.

Also, we're not sure whether our automated members can handle extensions
longer than 4 digits. I'd like to pass a substring of our
extension/destination_number in the call to Queue(). I couldn't find
documention of any Queue() option like this. Is it possible to control the
extension that the member receives?

Cheers,
David
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[asterisk-users] Getting DTMF from Asterisk Record?

[David Cunningham]

Hello,

Does anyone know if it's possible to get the non-terminating DTMF keys that
may have been pressed while using the Record command in Asterisk?

The intended purpose is that a caller can say a name or enter it using DTMF
keys. If we can get the keys in a variable that would be very helpful.

Thanks for any advice.

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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[asterisk-users] Compiling 15.2.0 and 15.2.1 Fails Others are Fine

[David Klaverstyn]

Hi All,

When 15.2.0 was released I tried to upgrade as I do when new versions are 
released but it failed to compile.  I figured it may be a bug so I waited for 
the next release but 15.2.1 also fails in the same location.  I can download, 
and compile 15.1.5 no problems at all.  I'm not sure if it is a 15.2.x problem 
or something else.

When I compile the following occurs which I could not find and answer for.

./libasteriskpj.so: undefined reference to `initBcg729EncoderChannel'
./libasteriskpj.so: undefined reference to `bcg729Decoder'
./libasteriskpj.so: undefined reference to `bcg729Encoder'
./libasteriskpj.so: undefined reference to `initBcg729DecoderChannel'
./libasteriskpj.so: undefined reference to `closeBcg729EncoderChannel'
./libasteriskpj.so: undefined reference to `closeBcg729DecoderChannel'


I am running this on a Raspberry Pi 3B, Raspbian 9.3 with Kernel 4.9.79-v7+ 
with all the latest updates.  I've been using the rPi for about four or so 
years now and have not experienced a problem like this one.

Any assistance will be greatly appreciated.

Regards
David.

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Re: [asterisk-users] Digium G100 and CID Dropping First Digit.

[David Klaverstyn]

On Mon, Jan 15, 2018 at 10:41:27PM +, David Klaverstyn wrote:
> port1 < Calling Number (len=12) [ Ext: 0  TON: National Number (2)  NPI: 
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> port1 <   Presentation: Presentation allowed of 
> network provided number (3)  '21xx' ]
> port1 < [70 0a c1 30 34 39 31 34 31 32 31 34]
> port1 < Called Number (len=12) [ Ext: 1  TON: Subscriber Number (4)  
> NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '049xx' ]
> 
> -- Accepting call from '21xx' to '049xx' on channel 0/5, span 
> 1

Don't know anything about the card you are using, but seeing ISDN signaling 
that the type of number (TON) is national and that means overhere that leading 
zeros are stripped, I see nothing wrong with it.
Looking at my old chan_dadhi configs there are options to prefix something 
based on TON. So over here I have configured:
nationalprefix = 0
to prefix the leading 0 for national numbers that callees expect.
The G100 manual contains the phrase "national prefix", but no info about it, so 
look into those prefix options.

--

I feel stupid now.  I should have figured that one out.  

On the Digium gateway: Configuration > T1/E1 > Advanced Signalling  > PRI 
Options : International, Nation and Local prefix.

Once I entered the correct digits International 00 and National 0, CID worked 
as expected.


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[asterisk-users] Digium G100 and CID Dropping First Digit.

[David Klaverstyn]

Hi All,

I have installed a number of Digium G100 devices in many countries like South 
Korea, Japan, Singapore and Australia.  I have just installed two in New 
Zealand and both sites are having a problem with Caller ID.  Incoming calls are 
dropping the first digit 0 from the caller ID.  I was previously using DAHDI 
and a TE121 device which may have been adding the 0, I'm not too sure about 
that.

Anyhow, is it possible the Digium G100 is causing the problem or would it be 
the Telco not passing the full CID number?  I have the latest firmware on the 
G100, the same with all my other locations.  I am using the latest Asterisk 
13.19.0.

The Telco is blaming the PBX for the problem so I'm hoping someone here can 
shed some light.  Below is a debug extract which I hope will help.

The 21 number should be 021.


Jan 16 10:53:16 G100-59-c4-ca asterisk[4674]: VERBOSE[4786]: 
chan_gtw/chan.c:4803 in dgm_pri_message: port1 < Calling Number (len=12) [ Ext: 
0  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) 
(1)
Jan 16 10:53:16 G100-59-c4-ca asterisk[4674]: VERBOSE[4786]: 
chan_gtw/chan.c:4803 in dgm_pri_message: port1 <   
Presentation: Presentation allowed of network provided number (3)  '21xx' ]
Jan 16 10:53:16 G100-59-c4-ca asterisk[4674]: VERBOSE[4786]: 
chan_gtw/chan.c:4803 in dgm_pri_message: port1 < [70 0a c1 30 34 39 31 34 31 32 
31 34]
Jan 16 10:53:16 G100-59-c4-ca asterisk[4674]: VERBOSE[4786]: 
chan_gtw/chan.c:4803 in dgm_pri_message: port1 < Called Number (len=12) [ Ext: 
1  TON: Subscriber Number (4)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) 
(1)  '049xx' ]

Jan 16 10:53:16 G100-59-c4-ca asterisk[4674]: VERBOSE[4786]: 
chan_gtw/sig_pri_new.c:5383 in pri_dchannel: -- Accepting call from 
'21xx' to '049xx' on channel 0/5, span 1

Thanks
David.
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Re: [asterisk-users] Blocking outgping caller id on a PRI E1

[David Duffett]

It is likely being set by your PRI provider.
Contact them to investigate.

All the best...

On 8 Nov 2017 9:03 am, "Neil Youngman"  wrote:

> I am trying to block/hide outgoing caller id on a PRI E1.
>
> It seems that it should be fairly simple, but it is defeating me.
>
> https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID says:
> "to hide your caller id, use: Set(CALLERID(num-pres)=prohib)"
>
> That doesn't seem to do it.
>
> The calls are originated from AMI and I have tried a blank "CallerId:"
> line and "CallerId: <>"". Neither of those made any difference.
>
> I have also tried "hidecallerid=yes" in chan_dahdi.conf, but that has also
> made no difference.
>
> I assume that I am missing something obvious?
>
> Neil Youngman
>
>
> Neil Youngman
> Developer
> Wirefast Limited
>
> Wirefast provides secure corporate messaging services.
> See our messaging solutions at  http://www.wirefast.com/
> Please consider the environment.
> Does this email or attachment need to be printed?
> This message contains confidential information and is intended only
> for the individual named. If you are not the named addressee you
> should not disseminate, distribute or copy this email. Please
> notify the sender immediately by email if you have received this
> email by mistake and delete this email from your system.
>
> Any views or opinions are solely those of the author
> and do not necessarily represent those of Wirefast Limited
>
> Email transmission cannot be guaranteed to be secure or error-free
> as information could be intercepted, corrupted, lost, destroyed,
> arrive late or incomplete, or contain viruses. The sender therefore
> does not accept liability for any errors or omissions in the contents
> of this message which arise as a result of email transmission.
> Wirefast Limited is registered in England & Wales
> Company number: 03865860
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>
> Wirefast definitions of classification can be found here:
> www.wirefast.com/classifications
>
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Re: [asterisk-users] TDM400P takes too long to ring

[David Duffett]

If you are trying to detect caller ID, and it is being supplied by the
telco in the format you have configured in /etc/chan_dahdi.conf then this
should not cause a delay. Are you actually seeing the caller ID being
displayed on the ringing phones?

If, however, the telco is not supplying caller ID info, or it is being
supplied in a format that you have not set up for, this is likely the cause
of the delay (looking for caller ID).

All the best,

David

On 27 April 2017 at 12:48, Ryan, Travis <ry...@oscarwinski.com> wrote:

> Hey all,
>
>
>
> I have a setup with two analog lines coming into and Asterisk 13 box with
> a TDM400P and it takes a lot of rings before asterisk takes over. I’ve
> traced this same box on two different analog providers so it probably isn’t
> a problem with them.
>
>
>
> I DO have callerid enabled and not sure I can turn it off (if they will
> let me). Any other ideas of making it ring through faster?
>
>
>
> By the time my internal phones get rang the customer has heard upwards of
> 7 rings. Some customers hang up thinking no one will answer.
>
>
>
> Also, I have fax detection off in my dialplan.
>
>
>
> Thanks,
>
> Travis
>
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*David Duffett*
Digium, Inc. · Director, Worldwide Asterisk Community
6 Landscape Close · Weston on the Green · Bicester · Oxfordshire OX25 3SX ·
UK
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[asterisk-users] Advice of Charge for non-Snom SIP phones

[David Cunningham]

Hello,

We are generating AOC messages via the AMI and trying to deliver them to
various brands of SIP phone.

When "snom_aoc_enabled = yes" in sip.conf then a message is set in the Snom
format correctly.

We're not having much luck sending any other type of AOC message though. I
can't find any documentation to say what if anything is available. The
"aoc_enable" setting doesn't seem to have any effect in sip.conf.

Can anyone advise if there is any other support for AOC over SIP besides
Snom, and how to configure it?

Thank you,

-- 
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
Australia: +61 (0) 2 8063 9019
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[asterisk-users] Custom INFO for Advice Of Charge

[David Cunningham]

Hello,

Does anyone know how to send a SIP INFO with custom data on a specified
channel from Asterisk? The intention is to provide an Advice Of Charge.

Asterisk has it's own AOC function but it only seems to support Snom's
format, not the more general XML format.

Searching came up with SIPSendCustomInfo but apparently it sends on all
active SIP channels, and is only available with TEST_FRAMEWORK.

Thanks for any advice.

-- 
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
Australia: +61 (0) 2 8063 9019
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