[asterisk-users] voicemails and recordings have words repeated

[Israel Gottlieb]

Hi all
i have run into a problem and cant seem to find the solution

calls that are recorded and lots of voicemails recorded you can her some of
the words repeated as if the person has said it twice
it happens by different callers
using pjsip on 18.9.0

any ideas?

thanks
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[asterisk-users] How to escape the & in BackGround (Dovid Bender

[Israel Gottlieb]

On Mon, Jan 17, 2022, 19:58  wrote:

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>1. )
>2. Re: How to escape the & in BackGround (Doug Lytle)
>3. Re: How to escape the & in BackGround (Dovid Bender)
>
>
> --
>
> Message: 1
> Date: Sun, 16 Jan 2022 14:19:28 -0500
> From: Dovid Bender 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: [asterisk-users] How to escape the & in BackGround
> Message-ID:
> <
> cam3tth3asg-ed-dmy7rnrwora18ldfpndqxtmwtfsp_ahns...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi,
>
> I am trying to play a sound file from AWS S3. The URL is something like
> this http://example.org?foo=bar=b. The issue seems to be that as soon as
> Asterisk see's the & it assumes there is a new file and the a=b is not sent
> along. I tried doing \& but that did not work. Does anyone know a way of
> telling Asterisk that & is part of the URL and to pass it along as a
> string?
> -- next part --
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> Message: 2
> Date: Sun, 16 Jan 2022 16:10:39 -0500
> From: Doug Lytle 
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] How to escape the & in BackGround
> Message-ID:
> 
> Content-Type: text/plain; charset="utf-8"; Format="flowed"
>
> On 1/16/22 2:19 PM, Dovid Bender wrote:
> > Does anyone know a way of telling Asterisk that & is part of the URL
> > and to pass it along as a string?
>
> Try enclosing the URL in single quotes,
>
> Doug
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> Message: 3
> Date: Sun, 16 Jan 2022 16:39:13 -0500
> From: Dovid Bender 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] How to escape the & in BackGround
> Message-ID:
>  f09d66jv0fbomkfbym...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> I tried single quotes, double quotes, backslash etc and none of it worked
>
> On Sun, Jan 16, 2022 at 16:11 Doug Lytle  wrote:
>
> > On 1/16/22 2:19 PM, Dovid Bender wrote:
> >
> > Does anyone know a way of telling Asterisk that & is part of the URL and
> > to pass it along as a string?
> >
> >
> > Try enclosing the URL in single quotes,
> >
> > Doug
> >
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> End of asterisk-users Digest, Vol 209, Issue 9
> **
>




URIENCODE?

>
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[asterisk-users] fax settings for audiocodes mp series fxs gateway and asterisk

[Israel Gottlieb]

Hi all

Does anyone have working settings for a audiocodes fxs gateway behind a
firewall to send faxes
thru asterisk not behind nat
i have tried multiple settings and haven't gotten it to work even partially
thanks,
israel
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[asterisk-users] please update contrib/scripts/get_mp3_source.sh to use https

[Israel Gottlieb]

contrib/scripts/get_mp3_source.sh

svn export has to be changed to https else it fails
thanks
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Re: [asterisk-users] DTMF rfc2833 missed when transfering to another server

[Israel Gottlieb]

well looks likes we solved it
the rtpkeepalive was set to 5 seconds on the trunk and every time asterisk
sends a rtpkeepalive a cn packet is sent
the same time a cn packet is sent asterisk loses the dtmf it was sent


On Wed, Dec 16, 2020 at 7:43 PM Israel Gottlieb  wrote:

> Hi all
> i have a asterisk server 16.11.1 (server A) that gets a call (leg A) and
> then calls a second server (leg B) server B is a freeswitch server
>
> the servers are configured all thru with rfc2833 for dtmf
> the caller enters a number a long 15 digit number like a credit card
> number or even a phone number and in alot of cases server B always
> doesn't get part of the digits from server A
>
> running a trace on server (A) i checked the trace of leg A and of leg B on
> the same server (A) and i see that the from the provider to asterisk has
> all digits correct but leg b going out the same server has a missed digit
>
> so either asterisk isnt getting all digits from the provider for some
> reason or it fails to regenerate the dtmf when sending to server b
> another think i noticed is asterisk generating a rtp  (cn) packet to leg
> every time it misses
>
> any idea how i can check what asterisk is seeing if its just sending the
> rtp without transcoding ?
> does anyone have a idea of what might be the problem
>
> using chan_sip
> rfc2833compensate=yes
> relaxdtmf=yes
>
> thanks
>
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[asterisk-users] DTMF rfc2833 missed when transfering to another server

[Israel Gottlieb]

Hi all
i have a asterisk server 16.11.1 (server A) that gets a call (leg A) and
then calls a second server (leg B) server B is a freeswitch server

the servers are configured all thru with rfc2833 for dtmf
the caller enters a number a long 15 digit number like a credit card number
or even a phone number and in alot of cases server B always doesn't get
part of the digits from server A

running a trace on server (A) i checked the trace of leg A and of leg B on
the same server (A) and i see that the from the provider to asterisk has
all digits correct but leg b going out the same server has a missed digit

so either asterisk isnt getting all digits from the provider for some
reason or it fails to regenerate the dtmf when sending to server b
another think i noticed is asterisk generating a rtp  (cn) packet to leg
every time it misses

any idea how i can check what asterisk is seeing if its just sending the
rtp without transcoding ?
does anyone have a idea of what might be the problem

using chan_sip
rfc2833compensate=yes
relaxdtmf=yes

thanks
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[asterisk-users] Fwd: blf problems after dialplan reload

[Israel Gottlieb]

Hi Guys
we have a system that uses a lot of custom hints based on the extension
the extensions use the format of ext-system for example 200-pbx01
when starting asterisk the "core show hints" show the correct hints and blf
works as expected

in the extensions.conf we have _.,hint,Custom:${exten}

when running dialplan reload all the hints lose the dashes (-) they become
200pbx01
of course blf doesnt work anymore

does anyone know a way to debug this as i think this is a asterisk bug and
would like to file a bug report
thanks,
israel
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Re: [asterisk-users] i sided recordings in asterisk 16.10

[Israel Gottlieb]

anthony answers inline

the funny thing is its like half half i have a recording of a clint
where the forst 8 minutes you hear 1 side and then you hear both until
the end of the conversation

i see they touched something in the last update

@jcolp

https://issues.asterisk.org/jira/browse/ASTERISK-28780

i do see alot of  asterisk notices in asterisk 16 alot

translate.c: 12547 lost frame(s) 12548/0 (slin@8000)->(alaw@8000)



On Tuesday 12 May 2020 at 12:28:51, Israel Gottlieb wrote:

>* Hi guys i upgraded to asterisk 16.10
*
>From what?  Did you change anything else at the same time?

asterisk 13.31

i have others with the same config on 13.33 and all is ok


>* and in most recordings you hear only leg A in the recording
*
What format are you recording with?

regular wav

>* sometimes you might hear a word of leg B
*
What dialplan command are you using to make the recording?

mixmontor

>* Did any body hit this problem?
*
Can you downgrade to your previous version to see whether the problem goes
away?

i will try tonight

What connectivity do you have on leg A and leg B - SIP phones?  Single SIP
accounts to an external provider?  SIP trunks for carrying multiple calls?
POTS hardware interface?

sip comes in on sip and back out


Regards,


Antony.

-- 
Schrödinger's rule of data integrity: the condition of any backup is unknown
until a restore is attempted.

   Please reply to the list;

please *don't* CC me.

On Tue, May 12, 2020 at 1:28 PM Israel Gottlieb  wrote:

> Hi guys i upgraded to asterisk 16.10 and in most recordings you here only
> leg A in the recording
> sometimes you might hear a word of leg B
> Did any body hit this problem?
> Thanks,
> israel
>
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[asterisk-users] i sided recordings in asterisk 16.10

[Israel Gottlieb]

Hi guys i upgraded to asterisk 16.10 and in most recordings you here only
leg A in the recording
sometimes you might hear a word of leg B
Did any body hit this problem?
Thanks,
israel
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[asterisk-users] no video when dialing between extension

[Israel Gottlieb]

hi all
im trying to call a door phone supporting video
i hear the audio but dont get video
i see this in the log
why should it try to translate?

Unable to find a codec translation path: (h264) -> (opus)

asterisk version 13.26
thanks for any help
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[asterisk-users] codec opus on centos 6 with asterisk 16

[Israel Gottlieb]

Hi list
does anyone know how i could use codec opus with asterisk 16 when using
centos 6
the prebuilt binary from digium doesnt load


Thanks,
Israel
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Re: [asterisk-users] Wanted: professional softphone

[Israel Gottlieb]

look at zoiper
oem.zoiper.com
you could create a url that creates a build with all credentials
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[asterisk-users] bridging in a 3rd caller without putting a caller on hold

[Israel Gottlieb]

hi all
How could conferance in a 3rd caller without put the second caller on hold
i would like to press a feature code mid call and have a 3rd caller enter
the call
this could be a real person or a automated system to take credit card info
mid call

thanks,
Israel
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Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1

[Israel Gottlieb]

how about sticking in a pbx between [c] and [h]
so when [h] hangsup you send to [s] if that is 3rd party else i dont see
how you could redirect [c] at all

else maybe ask them to have [h] redirect [c] to [s] then [h] will also be
out of the call

On Mon, Jul 1, 2019, 20:03  Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1. Re: Second Asterisk server SIP JOIN a call to conduct a
>   post-call survey (Joshua C. Colp)
>2. Re: Second Asterisk server SIP JOIN a call to conduct a
>   post-call survey (Jason N)
>3. Re: Second Asterisk server SIP JOIN a call to conduct a
>   post-call survey (Joshua C. Colp)
>
>
> --
>
> Message: 1
> Date: Mon, 01 Jul 2019 11:15:01 -0300
> From: "Joshua C. Colp" 
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
> to conduct  a post-call survey
> Message-ID: 
> Content-Type: text/plain;charset=utf-8
>
> On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote:
> > I am designing a solution for a hotel booking call center with the
> > following (mandatory) design: After the call from the customer with the
> > booking agent is complete (and the Hotel PBX disconnects from the
> > call), a second PBX takes over to conduct a survey of how the call
> > went. Both PBX’s are Asterisk based.
> >
> >
> > So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects,
> > the survey PBX [S] grabs the call and conducts the survey. [H] must
> > completely disconnect from the call before [S] can start the survey.
> > [H] cannot transfer/forward the call to [S].
> >
> >
> > At a high level the solution seems to be: On [C] connection to [H], [H]
> > sends call information to [S]. [S] issues a SIP JOIN to [C] and joins
> > the call. [S] somehow detects that [H] has disconnected and then begins
> > the survey.
> >
> >
> > Would the above work conceptually? If so, how do I tell Asterisk [S] to
> > contact [C] and join the call already in progress? (I can get call info
> > from [H] to [S]).
>
> It would be easiest for H to just Dial S after the first call leg is done.
> This can be done using the 'g' option to Dial[1] which continues dialplan
> application after the outgoing call leg hangs up. You could even send
> information as SIP headers if need be so S sees the info.
>
> [1]
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial
>
> --
> Joshua C. Colp
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
>
> --
>
> Message: 2
> Date: Mon, 1 Jul 2019 14:53:47 +
> From: "Jason N" 
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
> to  conduct a post-call survey
> Message-ID:
> <
> 0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000...@email.amazonses.com
> >
>
> Content-Type: text/plain;   charset="utf-8"
>
> Unfortunately I am not allowed any changes to H's PBX / dialplan.The
> restriction I have is that upon H's total disconnection from C, that S
> continues the call with C.  That's why I thought that if I could get S to
> SIP JOIN the call from C, that once H disconnects S can continue.   I can
> extract the SIP call info on H and pass that to S (so it can join the
> call).
>
> I'm just not sure if this concept is possible/practical.
>
>
> -Original Message-
> From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On
> Behalf Of Joshua C. Colp
> Sent: Monday, July 1, 2019 10:15 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call to
> conduct a post-call survey
>
> On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote:
> > I am designing a solution for a hotel booking call center with the
> > following (mandatory) design: After the call from the customer with
> > the booking agent is complete (and the Hotel PBX disconnects from the
> > call), a second PBX takes over to conduct a survey of how the call
> > went. Both PBX’s are Asterisk based.
> >
> >
> > So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects,
> > the survey PBX [S] grabs the call and conducts the survey. [H] must
> > completely disconnect from the call 

Re: [asterisk-users] Voicemail asking for login

[Israel Gottlieb]

 Does he have the same voicemail context?From: p...@fiberphone.co.nzSent: April 18, 2017 9:43 AMTo: asterisk-users@lists.digium.comReply-to: asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Voicemail asking for login  Hi D'ArcyOn 18/04/2017, at 5:17 am, D'Arcy Cain  wrote:One user (that we know of so far) has a different experience.  In that case they are asked for a mailbox number first.  I have tried searching for this issue but nothing seems to apply.  Most discussions are about "*97" vs. "*98".  Can anyone suggest another field of enquiry?Try this:	asterisk -r	core set verbose 10	[get user to trigger fault]	[examine console output, and post to list if still unclear]If you don't solve it yourself, then we'll be able to help further once we've seen the output.HTH,Pete-- 
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Re: [asterisk-users] multiple outbound invites

[Israel Gottlieb]

Maybe your firewall is blocking receiving packets from that provider or some 
sip helper is messing the returning packets so asterisk is not recieving a 
response and resending the invite

  Original Message  
From: j...@jeff.net
Sent: February 22, 2017 7:57 PM
To: asterisk-users@lists.digium.com
Reply-to: asterisk-users@lists.digium.com
Subject: [asterisk-users] multiple outbound invites


Hello,

I have two upstream providers we use for US termination.  The dialplan 
sends calls out the "primary" and if that fails for specific reasons, it 
sends the same call out the "secondary". This has worked well for us 
when we are lazy about keeping balances up, for example.

Starting a few days ago ALL calls sent to the 'primary' were returned as 
busy, though the secondary terminated them fine.  We have a balance, and 
funny enough international calls are going through fine, just not US 
calls.  I opened a ticket.

The response form the carrier is that our asterisk is sending four 
simultaneous invites within one second, and for that reason the call is 
rejected.

I did a packet trace and was able to confirm this is true - only US 
calls sent to this carrier cause our end to send four identical 
simultaneous invites.  When it fails, a single invite for the same call 
is sent to the secondary, which is terminated without issue.

Happy to send the SIP trace if any would care to see it, but is there a 
reason anyone can think of that our asterisk (11.11.0) would suddenly 
start doing this?  It may be that it has been doing it all along, and 
our carrier just started rejected calls that come in this way, I'm not sure.

Cheers,

j


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Re: [asterisk-users] First SIP-registering succeeds, second doesn't

[Israel Gottlieb]

 Disable all sip alg/helpers in the router


  Original Message  
From: andregronwal...@gmail.com
Sent: February 13, 2017 6:40 PM
To: asterisk-users@lists.digium.com
Reply-to: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] First SIP-registering succeeds, second doesn't

Some further information:
asterisk version: 13.13.1, pjsip (pjproject) 2.5.5


regards,
andre

Am 13.02.2017 um 17:32 schrieb Andre Gronwald:
> Hi all,
> I have a strange issue, with a some kind complicate architecture...
> A router of our internet provider is in front of another bintec rs353j 
> router, at which my freepbx installation is located. However, NAT etc. 
> seems to work fine.
> BUT: Something is not working...:
> When registering my sip-trunk towards my provider (3 different 
> providers, all behave comparable), everything works at first. Calls 
> are possible. But after some time, when the next REGISTER happens, the 
> answer of my provider is sent towards the wrong port. My freePBX 
> listens on 55060, where the first registration request are answered as 
> they should. in the second registration request wrong ports are used. 
> besides this, Header "Expires" is set to "0" and no "Allows" are 
> listed...
> [...]

> Any suggestions how to fix this? Or at least any idea what causes this?
>
> regards,
> andre


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Re: [asterisk-users] Call List Campaign to an IVR

[Israel Gottlieb]

there are providers which let you call directly to voicemail by using a
prefix

On Mon, Feb 6, 2017 at 8:28 PM, Tech Support 
wrote:

> I remember doing the testing and two calls going out at the same time
> don’t actually have to go out at the *exact* same time. The remote end will
> pick up one of the two calls, but there is no guarantee which one it will
> be. Also, if you let the first call ring too long, yes, the second call
> will go to voicemail,  but the first call will start ringing, which is
> something we wanted to avoid.
>
> John
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *On Behalf Of *Matt Riddell (lists)
> *Sent:* Monday, February 06, 2017 12:32 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Call List Campaign to an IVR
>
>
>
> Not really, doing the way below you don't even have to worry about it.
> They both go out at the same instant and as soon as it hits voicemail it
> disconnects the other leg.
>
>
>
> If you wanted you could leave it ringing for twenty minutes and it would
> still have the same effect.
>
> Kind regards,
>
>
>
> Matt
>
>
> On Feb 6, 2017, at 12:29 PM, Tech Support 
> wrote:
>
> That's the basics, but you have to nail the timing just right. The timing
> is
> really important to do it the right way.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com
> ] On Behalf Of Steve Edwards
> Sent: Monday, February 06, 2017 12:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Call List Campaign to an IVR
>
>
>
> On Mon, 6 Feb 2017, Tech Support wrote:
>
>
>
>  We were able to develop a feature to send the call to voicemail
>
> about 90% of the time. That way, an end user could (1) not be bothered by
> having to answer the call, (2)
>
>  delete the message without listening to it, or (3) listen to the
>
> message when it was most convenient for them. That way, they were in
> control
> and things were done on
>
>  their terms.
>
>
>
> On 6/02/2017, at 11:34 AM, Steve Edwards 
>
> wrote:
>
>
>
> Love the idea. How?
>
>
> On Mon, 6 Feb 2017, Matt Riddell wrote:
>
>
> exten =>
>
> _X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3)
>
>
> Amazing. Who knew?
>
> So how/why does this work?
>
> I see 2 calls going out to my cell. Does the first 'busy out' my number at
> my cell provider so the second goes straight to VM? What part does the
> '0111' play?
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867
> <(760)%20468-3867> PST
> https://www.linkedin.com/in/steve-edwards-4244281
>
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Re: [asterisk-users] How to send SIP_NOTIFY messages with variable content ?

[Israel Gottlieb]

snom could get lots of configuration options thru sip notify
i once tried updateing the display name on hot desking but ran in to his
problem of having to add it to sip conf staticly

On Wed, Jan 18, 2017 at 5:13 PM, Mark Wiater 
wrote:

>
> On 1/18/2017 9:58 AM, Tech Support wrote:
>
> For reconfiguring SIP phones? Can you give an example or short explanation?
>
>
> One can send a SIP notify with a check-config to the phone and have the
> phone re-download it's configuration files from a provisioning server.
>
> In the CLI, you can do a SIP NOTIFY with one of
>
> cisco-check-cfg, grandstream-check-cfg,  polycom-check-cfg,
> sipura-check-cfg, snom-check-cfg
>
> and the extension.
>
> I've done it with Yealink phones too, don't have the proper syntax in
> front of me though.
>
> Mark
>
>
>
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Re: [asterisk-users] Asterisk 11.24.1 garbled audio

[Israel Gottlieb]

Why not just timing test
It shows the timer used

On Nov 16, 2016 8:13 AM, "Stefan Viljoen"  wrote:

> Date: Tue, 15 Nov 2016 17:52:07 +0100
> From: Olivier 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Asterisk 11.24.1 garbled audio
> Message-ID:
>  gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> >Hi,
>
> >How can I double check which timer is currently is use in a running system
> ?
> >core show settings doesn't tell anything, if I'm not mistaken.
>
> How about
>
> module show like timi
>
> ?
>
> I use DAHDI timing, and this command on my system it shows:
>
> module show like timi 
> Module Description  Use
> Count
> res_timing_dahdi.soDAHDI Timing Interface   147
> 1 modules loaded
>
> To get this in my modules.conf I specify (among other things):
>
> load => res_timing_dahdi.so
> noload => res_timing_pthread.so
> noload => res_timing_timerfd.so
>
> I'm using 1.8.32.3, but should apply to you as well.
>
> Using DAHDI timing as I have had extensive problems under a heavily loaded
> system with both pthread and timerfd timing.
>
> Not sure if this still applies on your more recent Asterisk version though,
> but anyway - this is how you should be able to see what timing is being
> used
> on a running system.
>
> Regards
>
> Stefan
>
>
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Re: [asterisk-users] Dial and start music on hold after timeout

[Israel Gottlieb]

Are you sending progress?

בתאריך 24 באוג׳ 2016 13:40,‏ "Saint Michael"  כתב:

> ​I have the same exact issue. I cannot push any sounds or even Playtones
> to the caller, unless the channel is answered, which is not possible for
> billing reasons.
> I am also using the Local channel & Dial(PJSIP/...).
> I think this is a bug in Asterisk 13. The Dial function has not answered
> yet, so the Local channel should be able to play anything to the caller,
> without answering, in parallel with Dial.
> Should I open a JIRA ticket?
>
>
>
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Re: [asterisk-users] Dial and start music on hold after timeout

[Israel Gottlieb]

Maybe try progress() instead of answer ()

בתאריך 23 באוג׳ 2016 7:19 PM,‏ "Jean Aunis" <jean.au...@prescom.fr> כתב:

> Thank you, I just tried your suggestion. Strangely, the announcement is
> played only if I try to dial a SIP peer which is not available (not
> registered to be more precise). If the SIP peer is available, I only get
> the ring tone, and never hear the announcement. Here is the dialplan (I had
> to add an Answer() before the Dial, otherwise the announcement is never
> played, even in the first case) :
>
> exten = 007,1,Answer()
> same  = n,Dial(SIP/foo/s@playme,40)
>
> [playme]
> exten = s,1,Ringing()
> same  = n,Wait(10)
> same  = n,Playback(/var/lib/asterisk/sounddir/announce,noanswer)
> When it is working, I can see the following output in the CLI, which is
> not there otherwise :
> -- SIP/x requested media update control 26, passing it to
> Local/s@playme-05be;1
>
> Otherwise, no error message, Asterisk tells he is playing the announcement
> but I don't hear it.
>
> Best regards
>
> Jean Aunis
>
> Le 23/08/2016 à 16:07, David Duffett a écrit :
>
> How about:
>
> exten => s,1,Dial(SIP/alice/555@delayed-announce,40)
>
> [delayed-announce]
> exten =>  555,1,Wait(20)
> same => n,Playback(myannouncement,noanswer)
> same => n,NoOP(Whatever else you want to do goes here)
>
> The 'noanswer' option on the Playback means that SIP/alice should continue
> to ring for the remaining 20 of the 40 seconds, as the Playback will not
> answer (terminate) the call.
>
> Don't forget AstriCon this year - www.astricon.net
>
> On 23 August 2016 at 12:52, Israel Gottlieb <isr...@gmail.com> wrote:
>
>> You could m and make a moh file that has ringing the first 30 sec and
>> then the anouncment
>>
>> בתאריך 22 באוג׳ 2016 7:19 PM,‏ "Jean Aunis" <jean.au...@prescom.fr> כתב:
>>
>> Thank you for the idea. The problem with RetryDial, is that it will
>>> cancel the first call, play the announce and then dial the SIP peer once
>>> again, so the telephone will display a missed call. I would prefer to do
>>> everything in a single call.
>>>
>>> Le 22/08/2016 à 17:57, John Kiniston a écrit :
>>>
>>> You could try using RetryDial() instead of Dial, It supports playing an
>>> announcement.
>>>
>>>
>>> On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis <jean.au...@prescom.fr>
>>> wrote:
>>>
>>>> Sorry, I forgot to write that the SIP peer must keep ringing while the
>>>> announcement is being played.
>>>>
>>>> Le 22/08/2016 à 17:42, John Kiniston a écrit :
>>>>
>>>> This seems like the obvious answer but maybe I'm misunderstanding the
>>>> question.
>>>>
>>>> exten => s,1,Dial(SIP/alice,20)
>>>>  same =>   n,Playback(myannouncement)
>>>>  same =>   n,NoOP(Whatever else you want to do goes here)
>>>>
>>>> On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis <jean.au...@prescom.fr>
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> I am searching a way to dial a SIP peer, and if it does not answer
>>>>> within 20 seconds, play an announcement to the caller. This means that the
>>>>> caller would hear a ring tone for 20 seconds, and only then hear the
>>>>> announcement if the callee did not answer.
>>>>>
>>>>> I know it is possible to do this with ARI, but in this particular case
>>>>> I do not want to use ARI. I would like to do this purely with dialplan and
>>>>> AGI scripts, but I cannot find a way. I have read about the "m" option of
>>>>> Dial application, but it starts the announcement immediately, whereas I
>>>>> would like to start it after 20 seconds of timeout.
>>>>>
>>>>> Does anybody have an idea ?
>>>>>
>>>>> Best regards,
>>>>>
>>>>> Jean Aunis
>>>>>
>>>>>
>>>>> --
>>>>> _
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>>>

Re: [asterisk-users] Dial and start music on hold after timeout

[Israel Gottlieb]

You could m and make a moh file that has ringing the first 30 sec and then
the anouncment

בתאריך 22 באוג׳ 2016 7:19 PM,‏ "Jean Aunis"  כתב:

> Thank you for the idea. The problem with RetryDial, is that it will cancel
> the first call, play the announce and then dial the SIP peer once again, so
> the telephone will display a missed call. I would prefer to do everything
> in a single call.
>
> Le 22/08/2016 à 17:57, John Kiniston a écrit :
>
> You could try using RetryDial() instead of Dial, It supports playing an
> announcement.
>
>
> On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis  wrote:
>
>> Sorry, I forgot to write that the SIP peer must keep ringing while the
>> announcement is being played.
>>
>> Le 22/08/2016 à 17:42, John Kiniston a écrit :
>>
>> This seems like the obvious answer but maybe I'm misunderstanding the
>> question.
>>
>> exten => s,1,Dial(SIP/alice,20)
>>  same =>   n,Playback(myannouncement)
>>  same =>   n,NoOP(Whatever else you want to do goes here)
>>
>> On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis 
>> wrote:
>>
>>> Hello,
>>>
>>> I am searching a way to dial a SIP peer, and if it does not answer
>>> within 20 seconds, play an announcement to the caller. This means that the
>>> caller would hear a ring tone for 20 seconds, and only then hear the
>>> announcement if the callee did not answer.
>>>
>>> I know it is possible to do this with ARI, but in this particular case I
>>> do not want to use ARI. I would like to do this purely with dialplan and
>>> AGI scripts, but I cannot find a way. I have read about the "m" option of
>>> Dial application, but it starts the announcement immediately, whereas I
>>> would like to start it after 20 seconds of timeout.
>>>
>>> Does anybody have an idea ?
>>>
>>> Best regards,
>>>
>>> Jean Aunis
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   http://www.asterisk.org/hello
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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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>>
>
>
>
> --
> 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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Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Israel Gottlieb]

Could you please write the problem your having and not the reason to the
problem
Maybe the reason is something else

בתאריך 8 באוג׳ 2016 17:25,‏ "Tammy Firefly"  כתב:

Hi All,

We have asterisk 11.23 running sip to vitelity and from there IAX trunks
split off to where they need to go.  We are having a problem getting
chan_sip to quit ignoring re-invites from Vitelity.  Our side ends up
sending a reinvite which their side & they do not support us sending a
reinvite.  Ive tried:

canreinvite=no which was supposedly replaced by:

directmedia=no

Can anyone shed any light on this matter?  I'd love to get this fixed.

There is no firewall on this machine at all.

Thanks
--Tammy

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Re: [asterisk-users] Original Callerid on transfer in asterisk 13

[Israel Gottlieb]

In 11 setting trustrpid sendrpid is enough that phone getting the tranfered
call shows the name and number of the caller and not the tranferer
In 13 the same shows the transferrs info

בתאריך 11 באוג׳ 2016 00:21,‏ "Matt Fredrickson" <cres...@digium.com> כתב:

> How are you attempting to view the original CallerId?
>
> Matthew Fredrickson
>
> On Wed, Aug 10, 2016 at 2:59 PM, Israel Gottlieb <isr...@gmail.com> wrote:
> > Hi
> > Is there any configuration change in asterisk 13.9.1 to show original
> > callerid on a transfer
> > In asterisk 11.21 it works as expected
> >
> > Thanks
> >
> >
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>
>
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> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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[asterisk-users] Original Callerid on transfer in asterisk 13

[Israel Gottlieb]

Hi
Is there any configuration change in asterisk 13.9.1 to show original
callerid on a transfer
In asterisk 11.21 it works as expected

Thanks
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Re: [asterisk-users] Removing mailbox and password prompt for voicemail

[Israel Gottlieb]

Hi
Could please show us your dialplan as you have it now and the lines in the
log on that call it would probably help a lot more

בתאריך 4 באוג׳ 2016 5:00 PM,‏ "Nabeel"  כתב:

> What happens when you dial "*98" from your own
>> phone.
>
>
> I get password prompt if a password is set, and no password prompt if no
> password is set.
>
>
> On 4 August 2016 at 14:36, D'Arcy J.M. Cain  wrote:
>
>> On Thu, 4 Aug 2016 14:03:39 +0100
>> Nabeel  wrote:
>> > I should add, a password is *always* asked if a password has been set.
>> > There isn't a way to bypass that.
>>
>> Then something is wrong.
>>
>> http://darcy.vex.net/star98.mp3
>>
>> --
>> D'Arcy J.M. Cain
>> System Administrator, Vex.Net
>> http://www.Vex.Net/ IM:da...@vex.net
>> VoIP: sip:da...@vex.net
>>
>
> I wonder if the difference could be due to different Asterisk versions or
> the use of ODBC storage. For me, no password is asked only if I don't set a
> password. In other cases, it does. But i'll keep testing.
>
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Re: [asterisk-users] how to read sip debug

[Israel Gottlieb]

Another nice sip packet is sngrep
Shows realtime the sip flows
But i think you have to chk the asterisk answer in the dialplan logic to
chk what context its hitting etc.
בתאריך 6 ביולי 2016 10:05 PM,‏ "Steve Edwards" 
כתב:

> On Wed, 6 Jul 2016, Victor Villarreal wrote:
>
> If you experience problems with inbound calls from a SIP trunk or
>> provider, you can type in Asterisk cli 'core set debug 3' and then 'sip set
>> debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP provider or from
>> where it is supposed to come call.
>>
>
> Another path to enlightenment is to use tcpdump to capture the packets to
> a file and then use wireshark.
>
> Wireshark has a 'Telephony' menu and a 'SIP Flows' menu item where it will
> list all of the SIP packets. You can sort by column to help locate the
> packet of interest.
>
> Once found, you can click on 'Flow Sequence' and it will pop up a window
> showing the 'dialog ladder' that includes that packet. As you click on each
> packet in the flow, the main wireshark window will re-position to that
> packet so you can examine it in detail.
>
> Also on the 'SIP Flows' window is a 'Play Streams' button. Kind of scary
> how easy this is...
>
> --
> 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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Re: [asterisk-users] Delay after Answer

[Israel Gottlieb]

Another thing i would check is encryption is disabled on the snom
בתאריך 8 ביוני 2016 10:07,‏ "Israel Gottlieb" <isr...@gmail.com> כתב:

> Are you using stun? I have seen that when using stun
> בתאריך 8 ביוני 2016 09:54,‏ "Faheem Muhammad" <faheem2...@gmail.com> כתב:
>
>>
>>
>> Are you sure *nslookup  *command is returning as expected?
>> Also check the output of the below command.
>> >> hostname && hostname -s && hostname -f
>>
>>
>> On Tue, Jun 7, 2016 at 11:54 PM, Brent Davidson <
>> br...@texascountrytitle.com> wrote:
>>
>>> Well, I thought I had the problem solved.  Ported everything over to
>>> PJSip and build RDNS records for the phones and the server, but I am still
>>> experiencing the problem on incoming calls.
>>>
>>>
>>> On 6/7/2016 1:00 PM, Faheem Muhammad wrote:
>>>
>>> I've faced the same issue. The issue was related to DNS, the reverse
>>> lookup query failure caused the delay around(7-9 seconds). The purpose of
>>> reverse lookup is to block IP Spoofing attacks.
>>>
>>> Regards,
>>> Faheem
>>>
>>> On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson <
>>> br...@texascountrytitle.com> wrote:
>>>
>>>> I am having an issue with a couple of phones where they ring, but there
>>>> is a long delay after the phone is picked up before the audio starts.
>>>>
>>>> My setup:
>>>>
>>>>- Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
>>>>- Server is CentOS 7
>>>>- Quad core CPU with 16GB Ram
>>>>- 2 Snom 300 phones.
>>>>- NO NAT.  Server and phone are on the same subnet with only a
>>>>gigabit switch between them.
>>>>- Digium TDM400 analog card with 2 incoming analog PSTN lines
>>>>
>>>> When a call comes in, the system answers, IVR plays, caller dials an
>>>> extension, Snom 300 rings, handset picked up.  Caller continues to hear
>>>> ringing for another 7 to 10 seconds.  Answerer hears a click, a quick burst
>>>> of audio, then silence, then another click and audio is engaged.
>>>>
>>>> I have tried both SIP and RTP debugging and there are absolutely no
>>>> messages indicating any timeout or retransmit.  I am at a total loss.  In
>>>> the past I've always been able to find an answer to issues like this on my
>>>> own, but this time I just don't know.  I was even beginning to suspect the
>>>> network switch might be bad, but pinging between the server and the phones
>>>> shows no packet loss and 0.969ms average response time.
>>>>
>>>> What am I missing*?*
>>>> Thanks,
>>>> Brent Davidson
>>>>
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>>>
>>>
>>>
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Re: [asterisk-users] Delay after Answer

[Israel Gottlieb]

Are you using stun? I have seen that when using stun
בתאריך 8 ביוני 2016 09:54,‏ "Faheem Muhammad"  כתב:

>
>
> Are you sure *nslookup  *command is returning as expected?
> Also check the output of the below command.
> >> hostname && hostname -s && hostname -f
>
>
> On Tue, Jun 7, 2016 at 11:54 PM, Brent Davidson <
> br...@texascountrytitle.com> wrote:
>
>> Well, I thought I had the problem solved.  Ported everything over to
>> PJSip and build RDNS records for the phones and the server, but I am still
>> experiencing the problem on incoming calls.
>>
>>
>> On 6/7/2016 1:00 PM, Faheem Muhammad wrote:
>>
>> I've faced the same issue. The issue was related to DNS, the reverse
>> lookup query failure caused the delay around(7-9 seconds). The purpose of
>> reverse lookup is to block IP Spoofing attacks.
>>
>> Regards,
>> Faheem
>>
>> On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson <
>> br...@texascountrytitle.com> wrote:
>>
>>> I am having an issue with a couple of phones where they ring, but there
>>> is a long delay after the phone is picked up before the audio starts.
>>>
>>> My setup:
>>>
>>>- Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
>>>- Server is CentOS 7
>>>- Quad core CPU with 16GB Ram
>>>- 2 Snom 300 phones.
>>>- NO NAT.  Server and phone are on the same subnet with only a
>>>gigabit switch between them.
>>>- Digium TDM400 analog card with 2 incoming analog PSTN lines
>>>
>>> When a call comes in, the system answers, IVR plays, caller dials an
>>> extension, Snom 300 rings, handset picked up.  Caller continues to hear
>>> ringing for another 7 to 10 seconds.  Answerer hears a click, a quick burst
>>> of audio, then silence, then another click and audio is engaged.
>>>
>>> I have tried both SIP and RTP debugging and there are absolutely no
>>> messages indicating any timeout or retransmit.  I am at a total loss.  In
>>> the past I've always been able to find an answer to issues like this on my
>>> own, but this time I just don't know.  I was even beginning to suspect the
>>> network switch might be bad, but pinging between the server and the phones
>>> shows no packet loss and 0.969ms average response time.
>>>
>>> What am I missing*?*
>>> Thanks,
>>> Brent Davidson
>>>
>>> --
>>> _
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>>
>>
>>
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>
>
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Re: [asterisk-users] variable to get waittime of caller exiting queue

[Israel Gottlieb]

thank you

On Wed, May 18, 2016 at 5:48 PM, Faheem Muhammad <faheem2...@gmail.com>
wrote:

> Israel,
> You can calculate the time diff by this dialplan snippet.
>
>
> ---
> exten =
> _X.,1,Set(callstarttime=${STRFTIME(${EPOCH},,%Y%m%d)}${STRFTIME(${EPOCH},,%H%M%S)})
> exten => _X.,n,Queue(queue1)
> exten =
> _X.,n,Set(callendtime=${STRFTIME(${EPOCH},,%Y%m%d)}${STRFTIME(${EPOCH},,%H%M%S)})
> exten =_X.,n,Set(diff=$[${calltime1} -${calltime}])
> exten=_X.,n,NoOp(diff)
>
> -
>
> Regards,
> Muhammad
>
>
> On Wed, May 18, 2016 at 5:05 PM, Israel Gottlieb <isr...@gmail.com> wrote:
>
>> Hi all
>>
>> Is there anyway i could get in the dialplan  the amount of time a caller
>> waited in the queue before exiting?
>>
>> Thanks
>>
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>
>
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[asterisk-users] variable to get waittime of caller exiting queue

[Israel Gottlieb]

Hi all

Is there anyway i could get in the dialplan  the amount of time a caller
waited in the queue before exiting?

Thanks
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Re: [asterisk-users] How is Queue avg holdtime and avg talktime calculated

[Israel Gottlieb]

Thanks
For reports i dont those numbers but i use it for a wallboard and would
like to know what timeperiod is taken into account when mesuring the moving
average?


On Wed, May 11, 2016 at 1:02 PM, Ishfaq Malik <i...@pack-net.co.uk> wrote:

>
>
> On 11 May 2016 at 10:59, Ishfaq Malik <i...@pack-net.co.uk> wrote:
>
>>
>>
>> On 11 May 2016 at 10:24, Israel Gottlieb <isr...@gmail.com> wrote:
>>
>>>
>>> Hi all
>>>
>>> How is avg hold time and avg talktime calculated and over long a period
>>> of time?
>>>
>>> Thanks,
>>> Israel
>>>
>>>
>> Hi Israel
>>
>> If you are referring to the output of the queue show  command
>> then this is the response I received when asking this question previously:
>>
>> "Welcome to business logic embedded into app_queue.  The issue with the
>> queue show command rendering stats, is what timeframe are the stats
>> aggregated over?  IIRC, the calculations are using a moving
>> average[1].
>>
>> [1] http://en.wikipedia.org/wiki/Moving_average;
>>
>> If you want to find an average over a fixed period of time, your best bet is 
>> analysing the queue log. We had to do this ourselves when implementing a 
>> Dashboard with figures for the day.
>>
>> We found the figures outputted by the queue show  command to be 
>> misleading.
>>
>> Regards
>>
>>
>> Ish
>>
>>
>>
>>
> You can find my previous query and responses here:
>
> http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/282395
>
>
>
> --
>
> Ishfaq Malik
> Department: VOIP Support
> Company: Packnet Limited
> t: +44 (0)161 660 2350
> f: +44 (0)161 660 9825
> e: i...@pack-net.co.uk
> w: http://www.pack-net.co.uk
>
> Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
> 37 Ducie Street
> Manchester, M1 2JW
> COMPANY REG NO. 04920552
>
>
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[asterisk-users] How is Queue avg holdtime and avg talktime calculated

[Israel Gottlieb]

 Hi allHow is avg hold time and avg talktime calculated and over long a period of time?Thanks,Israel

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Re: [asterisk-users] Voice recognition IVR Is it possible?

[Israel Gottlieb]

if i remember correctly nerdvittles has a article
"google speech recognition api asterisk" brings results

On Tue, Feb 23, 2016 at 11:56 PM, Frank  wrote:

> On Tue, 2016-02-23 at 17:06 +, Steve Howes wrote:
>
> > Google?...
>
> Yeah... searched "google voice recognition api asterisk", browsed though
> various results. Nothing helpful for a beginner, very confusing bla
> bla...
>
> Thanks anyway for your help.
>
> F.
>
>
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[asterisk-users] תשובה: Dialing a call back out on same SIP trunk as it came in

[Israel Gottlieb]

Try putting progress instead of answer


  הודעה מקורית  
מאת: Tony Mountifield
נשלח: יום רביעי, 25 בנובמבר 2015 08:14
אל: asterisk-users@lists.digium.com
השב ל: Asterisk Users Mailing List - Non-Commercial Discussion
נושא: [asterisk-users] Dialing a call back out on same SIP trunk as it came in

I have a puzzling situation, and would be grateful for any insight.

I have a dialplan that forwards an incoming call out to another
number via the same SIP trunk as it came in on. e.g.

[from-siptrunk]
exten => 0123456789,1,NoOp
exten => 0123456789,n,Dial(SIP/siptrunk/0987654321)

Now, if I use a different SIP trunk for the outbound call, than the
inbound call came on, the call is set up fine - the Answer signal from the
called party gets propagated back to the caller, and they can hear each
other.

But if the outbound SIP trunk is the same as the one the call came in on,
the caller doesn't hear any progress, and has no notification of when the
call was answered. Neither can the parties hear each other.

I have tried this on two different machines using two different SIP
providers.

However, if I change the above NoOp to be Answer(100), i.e. answer the
inbound call before placing the outbound Dial, the caller hears progress
and when the called party answers, they hear each other fine.

Of course, if the called party is busy, the caller just hears in-band
busy tone, as the caller's inbound call was already answered.

Can anyone explain why I need the Answer? It feels wrong that I should.

The siptrunk entry contains canreinvite=no and directmedia=no.

The version of Asterisk on these boxes is 10.5.1, if that's relevant.

Thanks for any insight!

Cheers
Tony

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Play: t...@mountifield.org - http://tony.mountifield.org

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[asterisk-users] תשובה: Update new IP address (move temporarily) for INVITE

[Israel Gottlieb]

  Use redirectמאת: Sam Basanנשלח: יום שני, 9 בנובמבר 2015 12:40אל: 'asterisk maling list 'השב ל: sba...@bluebe.netנושא: [asterisk-users] Update new IP address (move temporarily) for INVITEHello, How can I update asterisk to send back move temporarily with updated IP address to incoming INVITE.i.e, Incoming call from ITSP to server 1 with x DID and there is a need to update the ITSP that the specified x DID number is allocated in server 2. Thanks,Sam Basan

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[asterisk-users] תשובה: Single SIP User on multiple location

[Israel Gottlieb]

Using pjsip you can have multiple endpoints for each extension

  הודעה מקורית  
מאת: A J Stiles
נשלח: יום רביעי, 2 בספטמבר 2015 13:10
אל: asterisk-users@lists.digium.com
השב ל: Asterisk Users Mailing List - Non-Commercial Discussion
נושא: Re: [asterisk-users] Single SIP User on multiple location

On Wednesday 02 Sep 2015, Avanish Shahi wrote:
> Now I’m trying to solve following problem. I have a requirement that
> each employee should have SIP phone at home, SIP phone in office,
> cell phone with same user.
> 
> 
> I want all those 3 phones to be “one extension”. So, if someone calls
> our company number and dials my extension - I’d like 3 phones to ring
> at the same time.
> 
> 
> e.g. Extension 555 for all the places and when anyone dial the
> extension 555 then it should ring at all the places simultaneously and
> user can pick any extension as desired.


exten => 555,Dial(SIP/555/7555/G1/07x)
will dial 555 and 7555 on the SIP trunk, and 07x via a hardware 
telephony card, until one of them answers.

You probably will want to use an AGI script to look up in a database the 
mobile number associated with the extension number; it will keep the dialplan 
sane.

-- 
AJS

Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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[asterisk-users] תשובה: Asterisk how to setup alarm too many outgoing calls from same user

[Israel Gottlieb]

  You could use the group functionCreate the group by extension and check how many calls are in the groupIf it's more than you allow then have it send a emailמאת: Ishfaq Malikנשלח: יום רביעי, 8 ביולי 2015 12:22אל: Asterisk Users Mailing List - Non-Commercial Discussionהשב ל: Asterisk Users Mailing List - Non-Commercial Discussionנושא: Re: [asterisk-users] Asterisk how to setup alarm too many outgoing calls from same userOn 6 July 2015 at 15:27, Motty Cruz motty.c...@gmail.com wrote:Hello,
I would like to setup a mechanism to trigger an alarm if user is deal too many numbers within a very short period of time. Safeguard against users hacked accounts.

can someone help?

Thanks,
You could use fail2ban for this by adding your own filter string specific for that user. It would have the advantage of blocking further calls as well as alerting you by email.RegardsIsh-- Ishfaq Malik 
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street 
Manchester, M1 2JW
COMPANY REG NO. 04920552




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[asterisk-users] תשובה: Branch based on call volume

[Israel Gottlieb]

  Look at the group functionמאת: Michelle Dupuisנשלח: יום שבת, 27 ביוני 2015 23:36אל: Asterisk Users Listהשב ל: Asterisk Users Mailing List - Non-Commercial Discussionנושא: [asterisk-users] Branch based on call volume






Is there a simple way to getcall volume from a particular trunkwithin the dialplan (for conditional branching)? 


I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13








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[asterisk-users] תשובה: תשובה: Missed call

[Israel Gottlieb]

It looks like you are dialing a external # then that won't work


  הודעה מקורית  
מאת: Luca Bertoncello
נשלח: יום שישי, 5 ביוני 2015 19:02
אל: asterisk-users@lists.digium.com
השב ל: Asterisk Users Mailing List - Non-Commercial Discussion
נושא: Re: [asterisk-users] תשובה: Missed call

Israel Gottlieb isr...@gmail.com schrieb:

 At the end of the Command you could use options one of them is the c (not
 apital) which sends a cancel event to the phone
 http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

Shalom Israel,

unfortunately it does not work as expected...
I wrote:

exten = _0049351222,n,Dial(SIP/0049351222SIP/004935,,Rc)

both phones ring, but if I answer from one phone, the other one say 1 missed
call...

Any other idea?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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[asterisk-users] תשובה: תשובה: Accessing an account from more than one phone

[Israel Gottlieb]

Shalom to you too
So that's the way to go


  הודעה מקורית  
מאת: Luca Bertoncello
נשלח: יום שישי, 5 ביוני 2015 09:51
אל: Asterisk Users Mailing List - Non-Commercial Discussion
השב ל: Asterisk Users Mailing List - Non-Commercial Discussion
נושא: Re: [asterisk-users] תשובה: Accessing an account from more than one phone

Zitat von Israel Gottlieb isr...@gmail.com:

Shalom, Israel!

 Using chan_sip you need to create another ‎user aand then dial both

 Using pjsip you can connect 2 devices

Thank you. Unfortunately it seems that I don't have pjsip available as 
package on the OpenWRT where I installed Asterisk... :(

I'll create another user.

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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Re: [asterisk-users] תשובה: Missed call

[Israel Gottlieb]

At the end of the Command you could use options one of them is the c (not
apital) which sends a cancel event to the phone
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

On Fri, Jun 5, 2015 at 9:53 AM, Luca Bertoncello lucab...@lucabert.de
wrote:

 Zitat von Israel Gottlieb isr...@gmail.com:

  If you the c option in the dial command it will send answered
 else where sip message to the phone and most ip phones understand that
 The cell will always display a missed call


 I'm very sorry, but I can't understand what you mean...
 Could you explain, maybe with an example?

 Thanks

 Luca Bertoncello
 (lucab...@lucabert.de)


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[asterisk-users] תשובה: Missed call

[Israel Gottlieb]

  If you the c option in the dial command it will send answered else where sip message to the phone and most ip phones understand thatThe cell will always display a missed callמאת: David Duffettנשלח: יום שישי, 5 ביוני 2015 09:37אל: Asterisk Users Mailing List - Non-Commercial Discussionהשב ל: Asterisk Users Mailing List - Non-Commercial Discussionנושא: Re: [asterisk-users] Missed callOn some SIP phones it is possible to turn off the missed call notifications, but I am not aware of a way to do the same on any cell phones. 
On 5 Jun 2015 07:29, "Mehmet Avcioglu" meh...@activecom.net wrote:
There is no signal that is sent to display a missed call. Your cell phone does that. If it rings and is not answered it counts that as a miss. The only way to avoid it is to not ring it. So instead of simultaneous ringing you can do sequential.

--
Mehmet Avcioglu
meh...@activecom.net

 On Jun 4, 2015, at 11:21 PM, Luca Bertoncello lucab...@lucabert.de wrote:

 Hi list!

 I configured Asterisk to forward the incoming call for a number to both phones.
 I wrote that:

 exten = _0049351222,n,Dial(SIP/0049351222SIP/004935,,R)

 of course it works...
 Now the problem is, that when a phone get the call, on the other phone I get "1 missed call"...
 Is it possible to avoid that and signaling the other phone, that the call was not "missed"?

 Thanks a lot
 Luca Bertoncello
 (lucab...@lucabert.de)


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[asterisk-users] תשובה: Accessing an account from more than one phone

[Israel Gottlieb]

Using chan_sip you need to create another ‎user aand then dial both

Using pjsip you can connect 2 devices

  הודעה מקורית  
מאת: Luca Bertoncello
נשלח: יום שישי, 5 ביוני 2015 09:24
אל: ML, Asterisk users
השב ל: Asterisk Users Mailing List - Non-Commercial Discussion
נושא: [asterisk-users] Accessing an account from more than one phone

Hi again!

I'm thinking about using my mobile phone to receive (and send) calls 
when I'm not at home (for example in holiday).
I can make my Asterisk reachable from Internet, of course, or I can 
use a VPN, that's not the problem...

My question is: can I log in to the same account from more than one device?
If yes, I can just configure my mobile phone with the same login of my 
phone at home and all works as expected.
If not, I have to create another user and to forward all calls to this 
user, too...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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[asterisk-users] תשובה: Forward loop protection...

[Israel Gottlieb]

  We could probably parse the rdnis field to see if it that hop is on the list מאת: Kevin Larsenנשלח: יום רביעי, 3 ביוני 2015 01:11אל: Asterisk Users Mailing List - Non-Commercial Discussionהשב ל: Asterisk Users Mailing List - Non-Commercial Discussionנושא: Re: [asterisk-users] Forward loop protection... The loop checking is a bit more challenging
than that. If Bob 
 forwards to Fred and Fred forwards to Sue, all is well when Bob
and 
 Fred head out for a beer. A little later, we’re in deep doo-do0
when
 Sue forwards to Bob. 

 Could this possibly mean that any person who
has CF set should never
 be available as CF Destination. Simple db entry/check can have this
done.

That just goes to show that the problem can get complex
pretty quickly. Using the original example above, it might be that you
want to allow the Bob to Fred to Sue forwards, but only stop it if the
Sue to Bob link is established, thus creating the loop. I wonder if you
could do some kind of recursive check where you follow each forward and
if you ever come back around to a number you have already checked you know
there is a loop.

To reuse the example above, on the creation of the
Bob to Fred forward, the database is checked to see if Fred has any forwards.
He doesn't, so is at the end of the forwarding chain. Now Fred forwards
to Sue. Again, she is at the end of the chain, so it is allowed. When Sue
goes to forward to Bob, the check shows that Bob has a forward. Not a problem,
but we create a temporary list that has Sue's number in it. Then we check
the next stage of forwarding. Bob forwards to Fred. Fred's is checked against
our temporary list and doesn't match, so we are still good. Bob's number
is now added to the temporary list and we check the forward Fred has in
place. Fred forward's to Sue. We check Sue's number against the temporary
list and it does exist. Thus we have a loop detected and the forward can
now be denied.

I am guessing with the recursion involved you might
want to do the check outside of Asterisk and pass the result back in. I
will also state that I have not had to do this deep checking in the past,
so these are just some initial thoughts on how I would start approaching
the problem. Of course, this also assumes that Bob, Fred, and Sue are all
on the same phone system. If you don't have a shared database to look at,
the problem just got harder indeed.


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[asterisk-users] תשובה: Seeking advice about ISDN BRI Cards

[Israel Gottlieb]

  הודעה מקורית  
מאת: jg
נשלח: יום חמישי, 28 במאי 2015 12:18
אל: Asterisk Users Mailing List - Non-Commercial Discussion
השב ל: Asterisk Users Mailing List - Non-Commercial Discussion
נושא: Re: [asterisk-users] Seeking advice about ISDN BRI Cards


 Thank you all for valuable input,

 another question: when do I actually need the echo cancellation
 (hardware / on board /on module ) ?

It depends on your environment. If there are still analog devices in addition 
to VoIP, I'd say 
always, but Asterisk has a rudimentary echo canceller already on board. The 
Telcos use echo 
cancellers themselves, but it cannot hurt to have a hardware canceller on your 
BRI card.

Nowadays I see more problems with reverberation in connection with cheap 
speakerphones or simple 
mics and speakers on PCs, but that's a different story.

jh

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Re: [asterisk-users] odbc connection timeout varable

[Israel Gottlieb]

thanks for the reply

On Wed, Nov 12, 2014 at 8:03 PM, Matthew Jordan mjor...@digium.com wrote:

 On Tue, Nov 11, 2014 at 1:43 PM, Israel Gottlieb isr...@gmail.com wrote:
  well it should but this morning my database hosted at a remote location
 was
  down due to conditions at the remote site
 
  the question isnt if it should happen or not
 
  the questions is there a way for me to know that the odbc query retruned
  empty because of a connection timeout?
 
  in curl i could get that info is there a way also for odbc?
 

 To just answer your question:

 No, there is no channel variable for this, nor anything in Asterisk
 that does this automatically.

 You would need to have something else monitoring your databases, and
 use it to inform Asterisk of the failure.

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

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[asterisk-users] odbc connection timeout varable

[Israel Gottlieb]

Hi all
Does anyone know of a variable that i could check to see if the reason func
odbc didnt return results was because of a timeout error so i could play a
audio file about that

thanks
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Re: [asterisk-users] odbc connection timeout varable

[Israel Gottlieb]

On Tuesday, November 11, 2014, jg webaccounts...@jgoettgens.de wrote:

 Why are you concerned? ODBC reconnects automatically if necessary.

 jg

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I would like to tell callers the system is down as apposed to saying no
info in system when the system is up or saying the info in the system
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Re: [asterisk-users] odbc connection timeout varable

[Israel Gottlieb]

right but that is the problem and i was wondering if there is way for
asterisk to set a variable when that happens just like curl

On Tue, Nov 11, 2014 at 5:37 PM, jg webaccounts...@jgoettgens.de wrote:

 Unless of course the database server is not running at all for some reason.

 But that's not exactly an Asterisk problem.


 jg

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Re: [asterisk-users] odbc connection timeout varable

[Israel Gottlieb]

well it should but this morning my database hosted at a remote location was
down due to conditions at the remote site

the question isnt if it should happen or not

the questions is there a way for me to know that the odbc query retruned
empty because of a connection timeout?

in curl i could get that info is there a way also for odbc?

On Tue, Nov 11, 2014 at 9:33 PM, jg webaccounts...@jgoettgens.de wrote:

 It should not happen. I have a couple of Asterisk servers using the ODBC
 connection. I never ever had any problem with ODBC or the database. What
 database are you using?


 jg

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Re: [asterisk-users] Loud Ringers and paging systems...

[Israel Gottlieb]

if you use a papt2 or so spa2101 then you could have alert info set to
different lengths or styles of ringers

i use that in a dorm with phones and have the phones ring short rings at
night so it wont wake up the students


On Tue, Aug 5, 2014 at 10:24 PM, Kevin Larsen 
kevin.lar...@pioneerballoon.com wrote:

 Working on a paging system for one of my sites and running into something
 I can't believe is this hard. In one of the zones, they want to have three
 different extensions ring over the pa system, using it as a loud ringer.
 Now the paging system does have a loud ringer built in and I can easily
 have it do a simultaneous ring, but all of the extensions will sound the
 same over the loud ringer. Of course, we want them to have different rings
 over the pa system so that all three people don't have to check their phone
 every time it rings.

 So far, the only semi solution I am coming up with (short of buying three
 different loud ringers and wiring them into the paging system) is to have
 my dialplan generate a call file that will make a second call to the paging
 system and play out an audio file based on who we are doing the loud ringer
 for. This has the disadvantage that it isn't a true loud ringer as it will
 only play for however long I tell it to and it won't cut off if they answer
 the phone before the audio file finishes playing.

 Anyone have any suggestions about a better way to handle this? Really
 hoping there is an Asterisk dialplan solution as I don't want to triple my
 paging hardware just to add one tiny piece of functionality.

 Kevin Larsen
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Re: [asterisk-users] Get last dialed number in a context?

[Israel Gottlieb]

you could save the info in astdb for the last call per extension and then
pull it from there


On Tue, Jun 3, 2014 at 12:31 PM, Stefan Gofferje li...@home.gofferje.net
wrote:

 Hi,

 I would like to implement an auto-redial function in a context. The idea
 is about like this:

 Dial a number
 Hear busy
 Hangup
 Pick up again
 Dial a code like *123
 = jumps into a context which redials until callresult is not busy

 Maybe like this:

 [autoredial]
 exten = s,1,Set(number=${CHANNEL(lastdialed)})
 exten = s,2,Dial(SIP/${number}@account,60,g)
 exten = s,3,Wait(15)
 exten = s,4,GotoIf( [ ${DIALSTATUS} = BUSY ]?2)
 exten = s,5,Hangup

 For that I'd need to somewhere get the last dialed number from the
 channel/line I'm initiating the call from. Is something like this
 already implemented?

 -S

 --
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  //\   Reg'd Linux User #247167   | VCP #2263
  V_/_  Heckler  Koch - the original point and click interface



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[asterisk-users] error cant write to function ODBC_DEVICES

[Israel Gottlieb]

Hi all

asterisk 1.8.23

I have odbc all setup to mysql but cant figure out why the dialplan wont
write to the odbc function

fubc_odbc.conf

[DEVICES]
dsn=device-conn;dsn in res_odbc not odbc.ini
readsql=SELECT call.callNum, call.city, devices.callId, devices.id FROM
call INNER JOIN devices ON call.id = devices.callId WHERE deviceNumber = '${
SQL_ESC(${ARG1})}'

writesql=insert into voted (callId,callNum,city,deviceId,SerialNum,
serverResponse) values (${VAL1},${VAL2},${VAL3},${VAL4},${VAL5},${VAL6}


extension.conf

the relevant line


same = n,set(ODBC_DEVICES()=${callid},${call},1,${deviceid},${num},${
serverupdate})


when sending the values from the cli using odbc write it works ok
reading from the dialplan  works ok
i tried sending plain values without variables

but from the dialplan gives me a error  cant write to function ODBC
_DEVICES

happy to hear any ideas
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Re: [asterisk-users] Queue callers with Callback option without lose their place

[Israel Gottlieb]

http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback



On Fri, Jun 1, 2012 at 1:45 PM, Satish Barot satish4aster...@gmail.comwrote:

 I believe you want your caller to request for a callback while he/she
 waits in a queue and when your agents are free, you want to call him back
 and place in a same position in a Queue where he/she has left the Queue.

 There exists an ugly(!) way of doing this.

 (1)Set parameter 'context' in queues.conf to some real context available
 in your dialplan
 (2)Set 'setqueueentryvar' and 'setqueuevar' to yes in queues.conf
 (3)Set paramet 'periodic-announce' to a custom audio file name announcing
 to caller somethink like ..'To get a callback press any key any'.(This
 sends the caller into context set by 'context' parameter when s/he presses
 any key while waiting in a queue)
 (4)A variable 'QUEUEPOSITION' would give you a last position of caller in
 a queue. (You can get this variable in a context set by 'context'
 parameter. Store the value somewhere in Database)
 (5)When you think your Agents are free, Generate a callfile OR use AMI to
 call the caller who has requested a callback.
 (6)Once call is answered, send him to Queue application with 'position'
 parameter set to the value of 'QUEUEPOSITION' of caller from database.

 --Satish Barot

 On Thu, May 31, 2012 at 9:18 PM, equis software 
 equissoftw...@gmail.comwrote:

 Is there any option in Asterisk distribution of this?

 Thanks.

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Re: [asterisk-users] concurrent channels limit

[Israel Gottlieb]

are you by chance using the a2billing script?

On Mon, Apr 2, 2012 at 5:43 PM, Syco syco...@gmail.com wrote:

 No, I don't do transcoding, I've disabled all the codec except for the
 g729.
 But in my last test I've found out what is the problem (not yet how to
 solve it)
 I make all my calls through a php agi, this old script works well on
 asterisk 1.4 and I want to move on 1.8.
 Just for test I've created three different (simplest) scripts:
 1 - stream a file codified in g729
 2 - make some mysql queries and stream the file
 3 - make an http hit and stream the file
 I stream an audio file to create calls that last some minute and test also
 the audio quality, I don't know if there's a better way.

 Anyway, if I use one of this 3 agi (also randomly) I'm able to establish
 up to 2500 channels with a perfect audio.

 If I use my old agi I could establish just 74 channels. I'm going mad on
 this because the number is not variable, is not one time 80 and the other
 70 and sometimes 88, it's always 74.
 The old agi script is a little longer than my test scripts, but it make
 the same things.
 I could accept the loss of some channels, but from 2500 to 74 there is a
 difference a little too big.



 On 02/04/2012 00:37, Matt Riddell wrote:

 How many g729 licenses do you have?  You sure you're not transcoding?


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Re: [asterisk-users] Streaming musiconhold via mpg123

[Israel Gottlieb]

that bug is running since the start of 1.8 and has been fixed in 1.8.9

https://issues.asterisk.org/jira/browse/ASTERISK-17474

i know it says that after the first time asterisks starts it works but
thats true only if the moh was loaded before the timing

its a long story but the fix is finally in

when typing timing test in the cli what timer to get if its dahdi then
thats the probably problem



On Wed, Feb 22, 2012 at 1:34 AM, Stephen Brown stephen.brow...@gmail.comwrote:

 On 2/21/2012 3:38 PM, isr...@gmail.com wrote:
  There is a bug in up to version 1.8.9 with external moh sources and
 dahdi timers

 Do you have a link to the bug report? I was unable to find anything but
 it's possible I'm not looking hard enough ;)

  Share with us your musiconhold.conf configuration please.

 Here it is... please excuse the mess, it's been a wild ride so my
 formatting/commenting has been left in-tact:

 ;
 ; Music on hold class definitions
 ; This is using the new 1.2 config file format, and will not work with 1.0
 ; based Asterisk systems
 ;
 ; #include musiconhold_custom.conf
 ; #include musiconhold_additional.conf
 ;[default]
 ;mode=custom
 ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r
 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074
 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0
 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074

 [test]
 mode=custom
 ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r
 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074
 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0
 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074
 application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000
 /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3

 I setup a simple 2 digit extension to call the test context and my MP3
 file nor my stream will play, and here's something else interesting: If
 use the MP3Player application to play an MP3, mpg123 spawns and plays
 it. I came to this conclusion by running ps aux | grep mpg while the
 song was playing.

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Israel Gottlieb]

wow i just tried in hebrew and i'll say just 1 word WOW

On Wed, Jan 4, 2012 at 9:48 PM, sean darcy seandar...@gmail.com wrote:

 On 1/4/2012 2:26 PM, Lefteris Zafiris wrote:


 Works beautifully. Amazing job Lefteris. Thanks.

 The best result I got in probability was 0.9725632 by saying, hello. I
 think there is some non-phonetic logic built-in as well. I tried, 1, 2
 and
 I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got,
 0.97256315. Probably Google sees the pattern?!

 What are some of the other tricks (if any) or consideration that one
 should
 make while creating a strong speech recognition enabled IVR?


 Google accepts sound files at any sampling rate (up to 44.1kHz) so if
 you can use some wideband codec ( eg g722)
 It can greatly improve the sound quality and the detection rates. For
 now the script supports 8kHz and 16kHz sampling rates
 for recording and it can be set by editing the scripts user defined
 parameters ( the variable $samplerate).
 Anything that improves the recording sound clarity will help, a good
 phone, low background noise level etc.
 I have also read that normalizing the recording and setting the gain
 to -5 db improves detection rates. I m experimenting with this at the
 moment and there will be some new code soon (as soon as i get sox
 working in RHEL/Centos 5 :P ).


 This is really spectacular. Thanks.

 I'm running Fedora 15, so I can use flac or sox. Any reason to prefer one
 over the other?

 sean




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Re: [asterisk-users] Not able to play wav files in asterisk

[Israel Gottlieb]

On Mon, Dec 26, 2011 at 9:03 AM, Steve Edwards asterisk@sedwards.comwrote:

 On Mon, 26 Dec 2011, isr...@gmail.com wrote:

  Rename the wav to ulaw
 Miss_audio.ulaw


 Very bad advice.


that might be but if you take a pcm ulaw encoded file and name it  .wav
asterisk will throw that error
I think asterisk should get smarter and read the header to get the format
or whatever else is needed and not only the extension of the filename

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000


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Re: [asterisk-users] using variables in the shell function

[Israel Gottlieb]

On Wed, Sep 14, 2011 at 5:27 AM, Dale Noll dn...@wi.rr.com wrote:

 On 09/13/2011 07:49 PM, Israel Gottlieb wrote:

 is it possible to pas variables to the shell function

 Set(recordingavail=${SHELL(ls
 /var/lib/asterisk/sounds/**custom/${TOPMENU})})

 im trying to see if a file is available before playing the file

 or does anybody have a different idea but not using agi

 asterisk 1.6.2.20
 thanks


 You should check out the STAT function.

 core show function STAT



 This should evaluate to 1
  ${STAT(e,/var/lib/asterisk/**sounds/en/vm-goodbye.gsm)})

 This should evaluate to 0
  ${STAT(e,/var/lib/asterisk/**sounds/en/xyzzy.gsm)}


 Dale



Thanks never noticed that function



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Re: [asterisk-users] using variables in the shell function

[Israel Gottlieb]

On Wed, Sep 14, 2011 at 4:08 AM, Steve Edwards asterisk@sedwards.comwrote:

 On Wed, 14 Sep 2011, Israel Gottlieb wrote:

  is it possible to pas variables to the shell function

 Set(recordingavail=${SHELL(ls /var/lib/asterisk/sounds/**
 custom/${TOPMENU})})

 im trying to see if a file is available before playing the file

 or does anybody have a different idea but not using agi


 Why not AGI?

 They both ('shelling out' or calling an AGI) have the same 'impact' on
 system resources.

 You can even write an AGI in shell if you lack the skills for other
 languages like C, PHP, or Perl.

 You should be able to cobble up an AGI in PHP (or Perl, but I'm not much of
 a Perl coder myself) just by cutting and pasting from some of the examples
 on voip-info.org.

 This simple task would be a great way for you to 'get your feet wet.'

 What will you do if the file is not available?


i know it has the same impact
im using asterisk since 0.4 yup lots of years mostly for the hobby and i
have written some stuff in bash and php  but never really got connected to
real programming. i read code no problem and understand exacttly what its
doing and do fix up what i need but most of the time just dont have the
patience to do it myself  (i have full time work in IT)

in this instance i was just writing a simple ivr maybe 10 lines were you
could descend down the ivr as long as there is a recording and if it could
be done simple within the dialplan i prefer that

i have made sophisticated ivr with lots up webservices and databases but
then i of course use a external language  (a function to speak to
webservices with xml parsing  i think would be a great addition to the
dialplan )

thanks
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[asterisk-users] using variables in the shell function

[Israel Gottlieb]

is it possible to pas variables to the shell function

Set(recordingavail=${SHELL(ls
/var/lib/asterisk/sounds/custom/${TOPMENU})})

im trying to see if a file is available before playing the file

or does anybody have a different idea but not using agi

asterisk 1.6.2.20
thanks
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Re: [asterisk-users] asterisk curl and utf8 problems

[Israel Gottlieb]

On Thu, Sep 8, 2011 at 1:52 AM, Israel Gottlieb isr...@gmail.com wrote:



 On Thu, Sep 8, 2011 at 1:47 AM, Israel Gottlieb isr...@gmail.com wrote:

 Hi all

 i have a very weird problem with curl and utf8 characters
 i'm trying to do a cnam lookup from a web-service with curl if the
 returned info is English or digits then the callerid name field gets
 populated with that but if the returned info is utf8 like Hebrew then the
 callerid field remains empty

 this is the curl statement
 exten = cnamlookup,1,Set(CALLERID(name)=${CURL(
 http://127.0.0.1:8080/GetCallerName?number=${CALLERID(num)})})


 any idea's?

 sorry for the missing info

 asterisk 1.8.6.0


 on asterisk 1.6 it was working and also on 1.2

 just a update

if i stick in a ascii char after the name like a period then it will work

seams like some bug to me
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[asterisk-users] asterisk curl and utf8 problems

[Israel Gottlieb]

Hi all

i have a very weird problem with curl and utf8 characters
i'm trying to do a cnam lookup from a web-service with curl if the returned
info is English or digits then the callerid name field gets populated with
that but if the returned info is utf8 like Hebrew then the callerid field
remains empty

this is the curl statement
exten = cnamlookup,1,Set(CALLERID(name)=${CURL(
http://127.0.0.1:8080/GetCallerName?number=${CALLERID(num)})})


any idea's?
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Re: [asterisk-users] asterisk curl and utf8 problems

[Israel Gottlieb]

On Thu, Sep 8, 2011 at 1:47 AM, Israel Gottlieb isr...@gmail.com wrote:

 Hi all

 i have a very weird problem with curl and utf8 characters
 i'm trying to do a cnam lookup from a web-service with curl if the returned
 info is English or digits then the callerid name field gets populated with
 that but if the returned info is utf8 like Hebrew then the callerid field
 remains empty

 this is the curl statement
 exten = cnamlookup,1,Set(CALLERID(name)=${CURL(
 http://127.0.0.1:8080/GetCallerName?number=${CALLERID(num)})})


 any idea's?

 sorry for the missing info

asterisk 1.8.6.0


on asterisk 1.6 it was working and also on 1.2
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Re: [asterisk-users] Increasing volume ?

[Israel Gottlieb]

Set(VOLUME(TX)=10) is correct but you arent putting it in a context so
asterisk doesnt know how to deal with it


do this

[bigbluebutton]
exten = _.,1,Set(VOLUME(TX)=10)
exten = _.,1,Set(VOLUME(RX)=10)
exten = _.,n,Goto(start-dialplan,s,1)
exten = _.,n,Hangup



On Thu, Aug 4, 2011 at 4:33 PM, Zeeshan Ali Shah zees...@infoshield.infowrote:

 any hint since it seems asterisk treat it as unknown directive


 On Thu, Aug 4, 2011 at 12:22 PM, Zeeshan Ali Shah zees...@infoshield.info
  wrote:

 but got these as well

 [Aug  4 12:21:08] WARNING[3082]: pbx_config.c:1588 pbx_load_config: ==!!==
 Unknown directive: Set(VOLUME(TX) at line 9 -- IGNORING!!!
 [Aug  4 12:21:08] WARNING[3082]: pbx_config.c:1588 pbx_load_config: ==!!==
 Unknown directive: SetGlobalVar(SetVOLUME(TX) at line 10 -- IGNORING!!!
 [Aug  4 12:21:08] WARNING[3082]: pbx_config.c:1588 pbx_load_config: ==!!==
 Unknown directive: SetGlobalVar(SetVOLUME(RX) at line 11 -- IGNORING!!!


 On Thu, Aug 4, 2011 at 12:20 PM, Zeeshan Ali Shah 
 zees...@infoshield.info wrote:

 Yes i tried the followings one by one with differnet values..
 Set(VOLUME(TX)=10)
 ;SetGlobalVar(VOLUME(TX)=10)
 ;SetGlobalVar(SetVOLUME(RX)=10)


  , but no improvement..  dont i have to change something in dialplan ?


 On Thu, Aug 4, 2011 at 11:17 AM, Matt Riddell li...@venturevoip.comwrote:

 On 4/08/11 9:16 PM, Zeeshan Ali Shah wrote:


 Tried below, but it still no improvement


 Zeeshan
 SetGlobalVar(VOLUME(TX)=10)
 SetGlobalVar(VOLUME(RX)=10)


 Have you tried just doing

 Set(VOLUME(TX)=10)

 and then 5 etc to make sure you are actually changing the volume?


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Re: [asterisk-users] FAX with SIP

[Israel Gottlieb]

On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.comwrote:

 On 07/21/2011 04:34 PM, Joaquin Sosa wrote:

 On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com  wrote:

 The magic sauce that you need is T.38 - Asterisk 1.6 supports this
 to a limited degree, and your ITSP will need to support it.

 The sip.conf.sample file and the voip-info wiki has all the
 information you need to try it out.


 Correct. However it would be helpful to note T.38 support in Asterisk
 is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and
 try to send a fax. It won't work!


 We do this in our testing all the time, and it works fine. Since you didn't
 specify any particular version of Asterisk, there's no way to associate your
 It won't work statement with anything in particular. Given the variations
 of T.38 implementations that exist in ATAs, carrier networks and other
 places, *any* T.38 connection that involves implementations from more than
 one vendor is (unfortunately) likely to have problems, whether any version
 of Asterisk is involved or not



well I tried  a linksys spa 8000 and 2102 thru
asterisk 1.8.3
1.8.4
1.6.2.16-19
sonus switch at itsp (012 israel)

and no luck
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Re: [asterisk-users] FAX with SIP

[Israel Gottlieb]

On Fri, Jul 22, 2011 at 12:50 AM, Kevin P. Fleming kpflem...@digium.comwrote:

 On 07/21/2011 04:43 PM, Israel Gottlieb wrote:



 On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.com
 mailto:kpflem...@digium.com wrote:

On 07/21/2011 04:34 PM, Joaquin Sosa wrote:

On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com
mailto:davies...@gmail.com  wrote:


The magic sauce that you need is T.38 - Asterisk 1.6
supports this
to a limited degree, and your ITSP will need to support it.

The sip.conf.sample file and the voip-info wiki has all the
information you need to try it out.


Correct. However it would be helpful to note T.38 support in
Asterisk
is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and
try to send a fax. It won't work!


We do this in our testing all the time, and it works fine. Since you
didn't specify any particular version of Asterisk, there's no way to
associate your It won't work statement with anything in
particular. Given the variations of T.38 implementations that exist
in ATAs, carrier networks and other places, *any* T.38 connection
that involves implementations from more than one vendor is
(unfortunately) likely to have problems, whether any version of
Asterisk is involved or not



 well I tried  a linksys spa 8000 and 2102 thru
 asterisk 1.8.3
 1.8.4
 1.6.2.16-19
 sonus switch at itsp (012 israel)

 and no luck


 We'd be happy to investigate why it failed, if you can capture the packet
 streams on both sides of Asterisk. Frequently, it's a configuration issue in
 at least one of the devices in the system.


NP I'll get that for you I have spent days trying to get it to work with no
luck


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 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Help: How can I Add my own Word in option packets in from field of SIP From Asterisk??

[Israel Gottlieb]

user-agent could be set in sip.conf

On Wed, Jul 20, 2011 at 12:43 PM, Alex Balashov
abalas...@evaristesys.comwrote:

 On 07/20/2011 05:00 AM, Masood Ahmed wrote:

  Hello All, Is there any one who can help me to change the From
 field parameters in option packets, I have seen that in option
 packtes asterisk sends its own information,If you see the below
 option packet i have highlighted the asterisk word in from field
 and in from field tag how can i changed it Please let me know same
 as in User Agent.


 These are internally generated, so there is no way to modify them without a
 source-level change.

 --
 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/

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[asterisk-users] multiple asterisk on 1 machine or other idea for using multiple network connection

[Israel Gottlieb]

Hi all

i have a scenario where i have 2 DSL lines (i know its not that reliable but
it fits the bill) connected to 1 box and would like my isp to round robin
between both dsl (to allow for more capacity - each dsl could get me thru
about 16-18 calls and i need about 30

incoming sip gets routed correct it goes out the same interface it came in
but rtp always gets routed out the default gateway
i can't force it with setting routes as i get the rtp for both links from
the same ip.

as far as i see its not possible for asterisk to have separate settings for
different contexts and thus being able to bind each context  to a differant
ip or differant static ip in sip contexts

so i am thinking of maybe setting up another asterisk instance on the same
box and binding that to the second ip and then routing from the second to
the first or setting up freeswitch as the second instance for the same idea

i have tried using opensbc and haven't gotten it to work the way i want


does anybody have a better idea?

also if i go with multiple asterisk how could i use the same binary and just
use a different  config (from Googling it looks as if it possible )

Thanks,
Israel
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[asterisk-users] setting sip headers when using call files

[Israel Gottlieb]

Hi

Does anybody have a idea how I could set sip headers when using call files?

I have to call out a specific trunk so I cant use local as the trunk

what i'm trying todo is send out calls as anonymous but at the itsp it
should be filed as being called out thru a specific DID and not the main DID
the provider has on file
for that I have to send the p-asserted but cant figure out how i would do
that with a call file

Thanks,
Israel
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Re: [asterisk-users] setting sip headers when using call files

[Israel Gottlieb]

On Thu, Apr 14, 2011 at 3:51 PM, Israel Gottlieb isr...@gmail.com wrote:

 Hi

 Does anybody have a idea how I could set sip headers when using call files?

 I have to call out a specific trunk so I cant use local as the trunk

 what i'm trying todo is send out calls as anonymous but at the itsp it
 should be filed as being called out thru a specific DID and not the main DID
 the provider has on file
 for that I have to send the p-asserted but cant figure out how i would do
 that with a call file

 Thanks,
 Israel



well responding to my own post i was just thinking that i could probably
create a separate context with the set and then dial using
Local/#@sip-add-headers (e.g,)

any other ideas?
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[asterisk-users] how to check if the call is using t38 except in the sip packets

[Israel Gottlieb]

How could i check if the call is using t38 except looking at the sip debug?

Is there any variable thats set or could be set?

thanks
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Re: [asterisk-users] spa8000 spa2102 t38 faxing

[Israel Gottlieb]

t38 from the itsp to asterisk using FFA =  works
t38 from the spa8000 to the local asterisk using FFA =  works   (no
differance if FAX Passthru Method is set to reinvite or nse)
g711  to a remote system =  works(i just turned off t38 on the
spa8000)

just the t38 pt is not working



the itsp is  using sonus switches

t38pt_udptl=yes,redundancy,maxdatagram=400  is set on the itsp

i couldnt yet get a normal trace as the system was very busy and lots of
debug info i'll try todo it tonight when the system is almost quiet

btw is there any variable to check if the channel is using t38 ?
Thanks








On Mon, Mar 28, 2011 at 1:10 AM, Larry Moore lmo...@starwon.com.au wrote:

 On 28/03/2011 5:48 AM, Israel Gottlieb wrote:

 still no luck
  i hear it change to t38 but it just doesnt connect


 Do you have two fax devices at your end, even a fax-modem attached to a
 computer will do?

 You are going to need to provide more information such as your current
 configuration and traces of the sessions.

 If you turn off all T.38 options in Asterisk and on the SPA you should
 still be able to make a transmissions using the G711 codecs.

 Can you confirm you are able to send a facsimile from your device using a
 PSTN line?


 Larry.

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Re: [asterisk-users] spa8000 spa2102 t38 faxing

[Israel Gottlieb]

still no luck
 i hear it change to t38 but it just doesnt connect


On Sun, Mar 27, 2011 at 5:26 AM, Larry Moore lmo...@starwon.com.au wrote:

 Perhaps this will help.

 I have a SPA8800 which has 4 x FXS  4 x FXO ports.

 It took me some time to produce a working configuration.

 In Asterisk I have the following where 904 is the extension of the
 fax-modem and itsp is you VoIP Service Provider.

 sip.conf

  [general]
  .
  .
  faxdetect=no
  t38pt_udptl=yes,redundancy,maxdatagram=400
  .
  .

  [904]
  ; Cisco SPA8800 FXS Port 4
  ; Analogue FAX Modem attached
  type=friend
  defaultuser=904
  secret=secret
  call-limit=2
  qualify=yes
  canreinvite=no
  directmedia=no
  directrtpsetup=no
  ignoresdpversion=yes
  transport=udp,tcp
  host=dynamic
  context=your_context
  faxdetect=no

  .
  .
  [itsp]
  .
  .
  faxdetect=yes
  ignoresdpversion=yes
  .
  .


 I am including information from my SPA8800 for one of the FXS ports I have
 a Fax Modem attached to, the key to getting it to work I believe is the FAX
 Tone Detect Mode.

 Audio Configuration

  Preferred Codec: G711a  Second Preferred Codec: Unspecified
  Third Preferred Codec: UnspecifiedUse Pref Codec Only: no
  Silence Supp Enable: yes  Silence Threshold: medium
  G729a Enable: no  Echo Canc Enable: yes
  G723 Enable: no  Echo Canc Adapt Enable: yes
  G726-16 Enable: no  Echo Supp Enable: yes
  G726-24 Enable: no  FAX CED Detect Enable: yes
  G726-32 Enable: no  FAX CNG Detect Enable: yes
  G726-40 Enable: no  FAX Passthru Codec: G711a
  DTMF Process INFO: yes  FAX Codec Symmetric: yes
  DTMF Process AVT: yes  FAX Passthru Method: ReINVITE
  DTMF Tx Method: AVT  DTMF Tx Mode: Strict
  DTMF Tx Strict Hold Off Time:  40FAX Process NSE: no
  Hook Flash Tx Method: None  FAX Disable ECAN: no
  Release Unused Codec: yes FAX Enable T38: yes
  FAX T38 Redundancy: 1  FAX Tone Detect Mode: callee only
  Symmetric RTP: yes

 Supplementary Service Settings

  CW Setting: noBlock CID Setting: no
  Block ANC Setting: noDND Setting: no
  CID Setting: yes  CWCID Setting: yes
  Dist Ring Setting: yesSecure Call Setting: no
  Message Waiting: noAccept Media Loopback Request: automatic
  Media Loopback Mode: sourceMedia Loopback Type: media

 Larry.

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[asterisk-users] spa8000 t38 faxing

[Israel Gottlieb]

Hi

I'm trying to get the spa 8000 used with a fax machine using t38 passthru
i have tried with 1.6.2 and 1.8.3 and is still a no go
the provider i use is 012 in israel wich supports t38 (i use it with ffa)

could anybody give me a clue how to get this working if it should

t38pt is set to yes in sip.conf

Thanks,
Israel
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Re: [asterisk-users] extend the timout on ringing for pri or sip

[Israel Gottlieb]

sorry i wasnt clear enough i meen inbound

On Thu, Feb 24, 2011 at 12:25 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:

 use the timeout option in the Dial application like so

 Dial(SIP/trunk,120)

 If you dont specify the timeout the default timeout used bya sterisk is
 probably more than 60 seconds.

 On Wed, Feb 23, 2011 at 3:17 PM, Israel Gottlieb isr...@gmail.com wrote:

 Hi

 Does anyone know how i could extend the timer for the ringing time on a
 pri or sip trunk ?
 Today the call gets a cancel request after a minute if not answerd yet
 is it on asterisk or is a provider side setting?


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 --
 Best Ragards
 Rizwan Qureshi
 VoIP/Asterisk Engineer
 Axvoice Inc.
 V: +92 (0)  6767 26
 E: rizwanhas...@gmail.com
 W: www.axvoice.com


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[asterisk-users] extend the timout on ringing for pri or sip

[Israel Gottlieb]

Hi

Does anyone know how i could extend the timer for the ringing time on a pri
or sip trunk ?
Today the call gets a cancel request after a minute if not answerd yet
is it on asterisk or is a provider side setting?
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[Asterisk-Users] How do I add a list of cidnames to the asterisk database in one shot ?

[Israel Gottlieb]

Hi! All


I have a table (currently access) of names and numbers which has aprox.
10,000 names  how could I get asterisk to look up the names in that database
or how do I enter them in to the asterisk database in one shot, if anyone
has a script please post (I remember one going around a while ago but I cant
seem to find it).

Thanks,
Israel Gottlieb


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