Re: [asterisk-users] moh reload not reloading/reading new musiconhold files

[Pete Mundy]

Hi Jonas

Wouldn't this do the job?

touch /etc/asterisk/musiconhold.conf ; asterisk -rx 'module reload 
res_musiconhold.so'

Pete


> On 31/03/2017, at 8:55 am, Jonas Kellens  wrote:
> 
> 
> I would not know how to automate this through script...
> 



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Re: [asterisk-users] CDR reporting solution

[motty cruz]

I installed CDR-Stats on Debian 8.7
http://cdr-stats.readthedocs.io/en/latest/installation/install-cdr-stats.html

I am trying to figure out how to import flat CSV file to CDR-Stats



On Wed, Mar 22, 2017 at 6:28 PM, Bruce Ferrell 
wrote:

> How about CDR to either MySQL or cdrlite and a quickie sql query? I could
> have added postgres, but I'm a DB bigot.  That would work too.
>
>
> On 03/22/2017 01:46 PM, Motty Cruz wrote:
>
>>
>> Hello, I am looking for CDR reporting solution? Any suggestions? I am
>> using Asterisk 13.13.1
>>
>> I would like a report on number of calls per extension.
>>
>> Thanks,
>> Motty
>>
>>
>>
>>
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Thanks for your support,
Motty
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Re: [asterisk-users] moh reload not reloading/reading new musiconhold files

[Jonas Kellens]

Hello

I can confirm that touch-ing /etc/asterisk/musiconhold.conf (just open 
with vi and close again) and then issuing a 'module reload 
res_musiconhold.so' on the Asterisk CLI makes the new files load into 
Asterisk.


Very strange !!

I would not know how to automate this through script...



Kind regards.


On 24-03-17 12:29, Daniel Journo wrote:


> Hello
> as you can read in my original post "moh reload" and "module reload 
res_musiconhold.so" does nothing.

> Only at restart the new files are available.
> Is this a bug ?? How can I get more debugging for this problem ??

I think there is currently a bug with MOH. For now, if you add a file 
to a moh folder, ‘touch musiconhold.conf’ and then reload moh.


Please let me know how it goes.

Kind regards

Dan Journo





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[asterisk-users] gcontacts to asterisk

[Atux Atux]

hello everyone. i am looking to automate the management of contacts to my
system (debian 8, with asterisk 11). at the moment i do create the astdb
with database put cidname.
I have searched a bit i have found the google contacts integration
https://zmonkey.org/blog/content/google-contacts-asterisk-caller-id
the problem is that it does not run at all.
my system also has mysql running that i could use.
has anyone any working solution for google contacts in the system, please?
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Re: [asterisk-users] Alphabet character in destination number (CDR)

[SamyGo]

Hi Ikka,

The last time I had this kind of problem the numbers in DB were altogether
different and the reason for that was inappropriate columns data-types in
DB.
You can also print out some CDR(${variable}) AFAIK in the dialplan and
verify that those are in original condition there or not.

Regards,
Sammy


On Thu, Mar 30, 2017 at 10:41 AM, Ikka Tirtawidjaja 
wrote:

> Dear all,
>
> I have PBX with asterisk 13.x
>
> a couple of IPPhone that connect to that asterisk PBX send an alphanumeric
> dialed phone number.
>
> for example, in my CDR table, field DST, it show dialed phone number like
> - 0C81318304632C  (it should be 081318304632)
> - 08D11157112 (it should be 0811157112).
>
> Why it's happening ? and how can I prevent it to happen ?
>
>
> Thanks in advance,
>
>
> Ikka
> Jakarta  - Indonesia
>
>
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[asterisk-users] Alphabet character in destination number (CDR)

[Ikka Tirtawidjaja]

Dear all,

I have PBX with asterisk 13.x

a couple of IPPhone that connect to that asterisk PBX send an alphanumeric
dialed phone number.

for example, in my CDR table, field DST, it show dialed phone number like
- 0C81318304632C  (it should be 081318304632)
- 08D11157112 (it should be 0811157112).

Why it's happening ? and how can I prevent it to happen ?


Thanks in advance,


Ikka
Jakarta  - Indonesia
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[asterisk-users] Asterisk 13.14.0. Debugging DTMF issues

[Olivier]

Hello,

I'm working on a (PJ)SIP trunking Asterisk machine with which I'm facing
issues with DTMF.
Installed version is 13.14.0.


1. In outbound calls SDP, I'm seeing these kind of lines:
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

I would expect events to range from 0 to 15, not to 16, as seen in rfc 4733
examples.
What is this event 16 for ?
Is there a way to configure this ?



2. With channel originated calls using Local prefix, I can read this on
console:
[2017-03-30 15:11:44] DTMF[11505][C-0056]: channel.c:4103 __ast_read:
DTMF begin '#' received on PJSIP/Foo-006f
[2017-03-30 15:11:44] DTMF[11505][C-0056]: channel.c:4114 __ast_read:
DTMF begin passthrough '#' on PJSIP/Foo-006f
[2017-03-30 15:11:44] DTMF[11501][C-0057]: channel.c:4103 __ast_read:
DTMF begin '#' received on Local/2@from-originate-0024;1
[2017-03-30 15:11:44] DTMF[11501][C-0057]: channel.c:4114 __ast_read:
DTMF begin passthrough '#' on Local/2@from-originate-0024;1
[2017-03-30 15:11:44] DTMF[11505][C-0056]: channel.c:4017 __ast_read:
DTMF end '#' received on PJSIP/Foo-006f, duration 180 ms

With pure inbound-outbound calls (calls coming in from PJSIP and leaving
through PJSIP), I get this:
[2017-03-30 15:51:01] DTMF[11650][C-006d]: channel.c:4017 __ast_read:
DTMF end '9' received on PJSIP/Bar-IPO-0093, duration 100 ms
[2017-03-30 15:51:01] DTMF[11650][C-006d]: channel.c:4044 __ast_read:
DTMF begin emulation of '9' with duration 100 queued on
PJSIP/Bar-IPO-0093
[2017-03-30 15:51:01] DTMF[11650][C-006d]: channel.c:4181 __ast_read:
DTMF end emulation of '9' queued on PJSIP/Bar-IPO-0093


Can I get both inbound and outbound DTMF on console ? How ?


3. Looking at DTMF duration (as logged by Asterisk console), I can see that
some (from mobile phone) have a 180ms duration while some, from an other
SIP trunk, have a 100ms duration.

Can I configure tone duration ? How ?
Should I configure this ?
Does this duration any real relation with the way a user presses keys ?


Best regards
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[asterisk-users] Logging endpoint IP address when PJSIP registration fails

[Sree Harsha Totakura]

Hi!

I am seeing a lot of warnings of these types:
res_pjsip_registrar.c: AOR '31' not found for endpoint 'anonymous'

I am guessing these are coming from a scanner trying to scan for the
extensions on the asterisk server.

Is there any way to print the IP address of the endpoint trying to
register an extension using PJSIP in asterisk 13?  I can then configure
fail2ban to temporarily hinder the scan.

Regards,
Sree

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