Re: [asterisk-users] Call files without permission for asterisk to read

[Rizwan Hisham]

Thanks for the responses.

Touching a file after setting permissions does not work. Asterisk only
looks at the new file only, not all the files in the directory.
Restarting asterisk does work, but dont want to do this.
Best way i think would be, as suggested by JG, to sync in a tmp directory
and at the time of switch-over mv to outgoing directory.

Cheers


On Fri, Nov 22, 2013 at 12:36 AM, Steve Edwards
wrote:

> On Thu, 21 Nov 2013, Rizwan Hisham wrote:
>
>  Hi all,I am syncing call files on my secondary asterisk server but
>> without permission to read for asterisk. So they should be executed when I
>> grant the right permissions (thats when my primary asterisk server crashes
>> or shutsdown somehow). But asterisk only tries to read the file at the time
>> of placing the file. So when i grant right permissions nothing happens. Is
>> there any workaround to this problem?
>>
>
> When you activate the secondary, 'touch' the files in the spool directory.
>
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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> _
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[asterisk-users] Call files without permission for asterisk to read

[Rizwan Hisham]

Hi all,
I am syncing call files on my secondary asterisk server but without
permission to read for asterisk. So they should be executed when I grant
the right permissions (thats when my primary asterisk server crashes or
shutsdown somehow). But asterisk only tries to read the file at the time of
placing the file. So when i grant right permissions nothing happens. Is
there any workaround to this problem?

I need to continue the execution of call files on secondary server if
primary server fails. The call files are suppose to retry for 45 mins if
the call does not get connected.

Thanks in advance.

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Rizwan H Qureshi

V: +971 (0) 528272154
linkedin.com/in/rhqureshi
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[asterisk-users] Realtime Call Files

[Rizwan Hisham]

Hi all,
Is there any way of originating calls in future without using call files?

We have 2 servers (1 active at a time). If we use call files with
modification date in future, on the 1st server and it dies and, we activate
the second server but we lose the call files.

I could have a cronjob on both servers and create callfiles reading
execution time from database, but this involves some other complications.

Any crazy ideas would be helpful.

Thanks

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[asterisk-users] Auto Redial Unconditional

[Rizwan Hisham]

Hi All,
I need a softphone (PC/Mobile) which does auto redial in any case
(noanswer,  answer, busy, congestion etc) after a given time interval. So
if the time interval was 5 secs, it would dial last number dialled after
every hangup (or every failure to dial).

Does anyone know such feature in a softphone?

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[asterisk-users] Confbridge Dynamic video_mode

[Rizwan Hisham]

Hi All,
I want to set the video_mode of the confbridge dynamically in the dialplan.
SO say if 5 users join the conference with follow_talker mode, it should
work like that (and it does). But if 6th user changes the video_mode to
first_marked and gets marked in the dial plan and joins the conference, he
does not become the single  video source of the conf. The video mode stays
follow_talker.

I have tried changing conf mode dynamically in dialplan but does not work
following is my settings and dialplan:

confbridge.conf
--
[common_bridge]
type=bridge
record_conference=yes
internal_sample_rate=auto
mixing_interval=20
video_mode = none
conference participants.
release_as_single_video_src

[default_user]
type=user
dsp_talking_threshold=128
dsp_silence_threshold=2000
talk_detection_events=yes

extensions.conf
---

exten => 200,1,Noop(Going to ConfBridge now)
same => n,SET(CONFBRIDGE(bridge,video_mode)=follow_talker)
same => n,Answer()
same => n,Confbridge(1234,common_bridge,,sample_user_menu)
same => n,NOOP(${CONFBRIDGESTATUS})


exten => 300,1,SET(CONFBRIDGE(user,marked)=yes)
same => n,SET(CONFBRIDGE(bridge,video_mode)=first_marked)
same => n,Answer()
same => n,Confbridge(1234,common_bridge,,sample_user_menu)
same => n,NOOP(${CONFBRIDGESTATUS})

Marked user dials 300 and all others dial 200.

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Re: [asterisk-users] Call Transfer not working

[Rizwan Hisham]

Thanks everyone. I was using the Tt flag but in the wrong place in the dial
application.

Cheers

On Mon, Apr 9, 2012 at 4:54 PM, Takehiro Matsushima <
takehiro.dream...@gmail.com> wrote:

> Thank you so much.
>
> OK, I understood that to transfer the call "t" is usually used, is it
> right?
> And I should write so in my last mail.
>
> "t" and "T" are described with same sentences in official wiki...
>
> Regards,
> Takehiro Matsushima
>
>
>
> 2012/4/9 Chris Bagnall :
> > On 9/4/12 3:04 am, Takehiro Matsushima wrote:
> >>
> >> // I don't know what's difference "t" and "T".
> >
> >
> > T allows the caller to transfer. t allows the called user to transfer.
> >
> > You very rarely want "Tt" - since I doubt you want an incoming caller to
> be
> > able to transfer their call all over the place. You usually want "t" on
> > incoming calls and "T" on outgoing calls.
> >
> > Kind regards,
> >
> > Chris
> > --
> > This email is made from 100% recycled electrons
> >
> >
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Rizwan Qureshi
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Axvoice Inc.

V: +92 (0)  6767 26
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[asterisk-users] Call Transfer not working

[Rizwan Hisham]

Hi All,
I am using asterisk 1.8.11 on centos 5. I have realtime sip peers with dtmf
setting rfc2833 and inband. I have also enabled blind and attended transfer
features in features.conf but still call transfers dont work. I have setup
transfer feature in past but i dont think i am missing anything this time.
I just dont have any clue why its not working. I have tried using ATAs and
softphones but cant make it to work. Can anyone help? Am I missing anything?

features show output:
===
Builtin Feature   Default Current
---   --- ---
Pickup*8  *8
Blind Transfer#   #1
Attended Transfer *2
One Touch Monitor
Disconnect Call   *   *
Park Call
One Touch MixMonitor
==
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Rizwan Qureshi
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Axvoice Inc.

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Re: [asterisk-users] Background music during a call

[Rizwan Hisham]

Its in testing phase right now Ioan. Also the curse thing is just an idea.
We may not implement it actually, or maybe we do in future but not now. Lets
see.

On Tue, May 10, 2011 at 10:23 PM, Ioan Indreias  wrote:

> Glad to know it works for you.
> I would like to hear your love/curse MOH - do you have some links to
> your mp3 files? :)
>
> BR,
> Ioan (with capital i)
>
> On Tue, May 10, 2011 at 6:59 PM, Rizwan Hisham 
> wrote:
> > Ooops,
> >
> > here is the correct version, Missed the capital X option in meetme before
> > which lets you control the volume etc.
>
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-- 
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Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.

V: +92 (0)  6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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Re: [asterisk-users] Background music during a call

[Rizwan Hisham]

Ooops,

here is the correct version, Missed the capital X option in meetme before
which lets you control the volume etc.

[chat-room]
exten => love,1,Goto(love-a,1)
exten => love,2,Goto(love-b,1)

exten => love-a,1,Set(__MOH=love)
exten => love-a,n,Dial(Local/fake@chat-
room,,G(chat-room,chat,1))

exten => love-b,1,Goto(chat,100)

exten => curse,1,Goto(curse-a,1)
exten => curse,2,Goto(curse-b,1)

exten => curse-a,1,Set(__MOH=curse)
exten => curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))

exten => curse-b,1,Goto(chat,100)

exten => fake,1,Answer
exten => fake,2,MusicOnHold(${MOH})

exten => chat,1,Goto(100)
exten => chat,2,MeetMe(${MM},dx1qX)

exten => chat,100,MeetMe(${MM},daAx1qX)

exten => h,1,MeetMeAdmin(${MM},K)

exten => 4,1,MeetMeAdmin(${MM},t,2)
exten => 6,1,MeetMeAdmin(${MM},T,2)
exten => 2,1,MeetMeAdmin(${MM},M,2)
exten => 8,1,MeetMeAdmin(${MM},m,2)

exten=> _X,2,Goto(chat-room,chat,100)


On Tue, May 10, 2011 at 9:57 PM, Rizwan Hisham wrote:

> Very nice Loan. Here is the chat-room dialplan with a little tweek which
> lets you set the volume up/down or mute/unmute the song.
>
> Use 4/6 to increase/decrease the volume and 2/8 to mute/unmute the song
>
>
> [chat-room]
> exten => love,1,Goto(love-a,1)
> exten => love,2,Goto(love-b,1)
>
> exten => love-a,1,Set(__MOH=love)
> exten => love-a,n,Dial(Local/fake@chat-
> room,,G(chat-room,chat,1))
>
> exten => love-b,1,Goto(chat,100)
>
> exten => curse,1,Goto(curse-a,1)
> exten => curse,2,Goto(curse-b,1)
>
> exten => curse-a,1,Set(__MOH=curse)
> exten => curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
>
> exten => curse-b,1,Goto(chat,100)
>
> exten => fake,1,Answer
> exten => fake,2,MusicOnHold(${MOH})
>
> exten => chat,1,Goto(100)
> exten => chat,2,MeetMe(${MM},dx1q)
>
> exten => chat,100,MeetMe(${MM},daAx1q)
>
> exten => h,1,MeetMeAdmin(${MM},K)
>
> exten => 4,1,MeetMeAdmin(${MM},t,2)
> exten => 6,1,MeetMeAdmin(${MM},T,2)
> exten => 2,1,MeetMeAdmin(${MM},M,2)
> exten => 8,1,MeetMeAdmin(${MM},m,2)
>
> exten=> _X,2,Goto(chat-room,chat,100)
>
> Here channel 2 always seem to be the one playing the MOH, thats why its
> hard coded into the MeetMeAdmin application.
>
> If there is a another way to know which channel is playing the song then
> please do let me know.
>
> Cheers
>
>
>
> On Tue, May 10, 2011 at 9:33 AM, Rizwan Hisham wrote:
>
>> Thanks a lot loan. Will try it today.
>>
>> Cheers
>>
>>
>> On Mon, May 9, 2011 at 6:25 PM, Ioan Indreias  wrote:
>>
>>> Updated dialplan: fix a typo when using MOH variable and now you have
>>> truly dynamic conference rooms.
>>>
>>> Have fun,
>>> Ioan.
>>>
>>> +
>>> exten => _[12]XXX,1,Set(__MM=${EPOCH})
>>> exten => _1XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,love,1))
>>> exten => _2XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,curse,1))
>>>
>>> [chat-room]
>>> exten => love,1,Goto(love-a,1)
>>> exten => love,2,Goto(love-b,1)
>>>
>>> exten => love-a,1,Set(__MOH=love)
>>> exten => love-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
>>>
>>> exten => love-b,1,Goto(chat,100)
>>>
>>> exten => curse,1,Goto(curse-a,1)
>>> exten => curse,2,Goto(curse-b,1)
>>>
>>> exten => curse-a,1,Set(__MOH=curse)
>>> exten => curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
>>>
>>> exten => curse-b,1,Goto(chat,100)
>>>
>>> exten => fake,1,Answer
>>> exten => fake,2,MusicOnHold(${MOH})
>>>
>>> exten => chat,1,Goto(100)
>>> exten => chat,2,MeetMe(${MM},dx1q)
>>>
>>> exten => chat,100,MeetMe(${MM},daAx1q)
>>>
>>> exten => h,1,MeetMeAdmin(${MM},K)
>>> +
>>>
>>> On Mon, May 9, 2011 at 4:02 PM, Ioan Indreias 
>>> wrote:
>>> > I have tested the following dialplan and it could be used as a
>>> > starting point. What you have to resolve is how to generate different
>>> > MeetMe conference room - in the example we have only one room = 1234
>>> >
>>> > If you prefix the dialled extension with 1 => you will have a "lovely
>>> > chat". With 2 -> "cursing chat".
>>> >
>>> > HTH,
>>> >
>>> > Ioan
>>

Re: [asterisk-users] Background music during a call

[Rizwan Hisham]

Very nice Loan. Here is the chat-room dialplan with a little tweek which
lets you set the volume up/down or mute/unmute the song.

Use 4/6 to increase/decrease the volume and 2/8 to mute/unmute the song

[chat-room]
exten => love,1,Goto(love-a,1)
exten => love,2,Goto(love-b,1)

exten => love-a,1,Set(__MOH=love)
exten => love-a,n,Dial(Local/fake@chat-
room,,G(chat-room,chat,1))

exten => love-b,1,Goto(chat,100)

exten => curse,1,Goto(curse-a,1)
exten => curse,2,Goto(curse-b,1)

exten => curse-a,1,Set(__MOH=curse)
exten => curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))

exten => curse-b,1,Goto(chat,100)

exten => fake,1,Answer
exten => fake,2,MusicOnHold(${MOH})

exten => chat,1,Goto(100)
exten => chat,2,MeetMe(${MM},dx1q)

exten => chat,100,MeetMe(${MM},daAx1q)

exten => h,1,MeetMeAdmin(${MM},K)

exten => 4,1,MeetMeAdmin(${MM},t,2)
exten => 6,1,MeetMeAdmin(${MM},T,2)
exten => 2,1,MeetMeAdmin(${MM},M,2)
exten => 8,1,MeetMeAdmin(${MM},m,2)

exten=> _X,2,Goto(chat-room,chat,100)

Here channel 2 always seem to be the one playing the MOH, thats why its hard
coded into the MeetMeAdmin application.

If there is a another way to know which channel is playing the song then
please do let me know.

Cheers


On Tue, May 10, 2011 at 9:33 AM, Rizwan Hisham wrote:

> Thanks a lot loan. Will try it today.
>
> Cheers
>
>
> On Mon, May 9, 2011 at 6:25 PM, Ioan Indreias  wrote:
>
>> Updated dialplan: fix a typo when using MOH variable and now you have
>> truly dynamic conference rooms.
>>
>> Have fun,
>> Ioan.
>>
>> +
>> exten => _[12]XXX,1,Set(__MM=${EPOCH})
>> exten => _1XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,love,1))
>> exten => _2XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,curse,1))
>>
>> [chat-room]
>> exten => love,1,Goto(love-a,1)
>> exten => love,2,Goto(love-b,1)
>>
>> exten => love-a,1,Set(__MOH=love)
>> exten => love-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
>>
>> exten => love-b,1,Goto(chat,100)
>>
>> exten => curse,1,Goto(curse-a,1)
>> exten => curse,2,Goto(curse-b,1)
>>
>> exten => curse-a,1,Set(__MOH=curse)
>> exten => curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
>>
>> exten => curse-b,1,Goto(chat,100)
>>
>> exten => fake,1,Answer
>> exten => fake,2,MusicOnHold(${MOH})
>>
>> exten => chat,1,Goto(100)
>> exten => chat,2,MeetMe(${MM},dx1q)
>>
>> exten => chat,100,MeetMe(${MM},daAx1q)
>>
>> exten => h,1,MeetMeAdmin(${MM},K)
>> +
>>
>> On Mon, May 9, 2011 at 4:02 PM, Ioan Indreias  wrote:
>> > I have tested the following dialplan and it could be used as a
>> > starting point. What you have to resolve is how to generate different
>> > MeetMe conference room - in the example we have only one room = 1234
>> >
>> > If you prefix the dialled extension with 1 => you will have a "lovely
>> > chat". With 2 -> "cursing chat".
>> >
>> > HTH,
>> >
>> > Ioan
>>
>> --
>> _
>> -- 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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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Best Ragards
> Rizwan Qureshi
> VoIP/Asterisk Engineer
> Axvoice Inc.
>
> V: +92 (0)  6767 26
> E: rizwanhas...@gmail.com
> W: www.axvoice.com
>
>


-- 
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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Re: [asterisk-users] Background music during a call

[Rizwan Hisham]

Thanks a lot loan. Will try it today.

Cheers

On Mon, May 9, 2011 at 6:25 PM, Ioan Indreias  wrote:

> Updated dialplan: fix a typo when using MOH variable and now you have
> truly dynamic conference rooms.
>
> Have fun,
> Ioan.
>
> +
> exten => _[12]XXX,1,Set(__MM=${EPOCH})
> exten => _1XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,love,1))
> exten => _2XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,curse,1))
>
> [chat-room]
> exten => love,1,Goto(love-a,1)
> exten => love,2,Goto(love-b,1)
>
> exten => love-a,1,Set(__MOH=love)
> exten => love-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
>
> exten => love-b,1,Goto(chat,100)
>
> exten => curse,1,Goto(curse-a,1)
> exten => curse,2,Goto(curse-b,1)
>
> exten => curse-a,1,Set(__MOH=curse)
> exten => curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))
>
> exten => curse-b,1,Goto(chat,100)
>
> exten => fake,1,Answer
> exten => fake,2,MusicOnHold(${MOH})
>
> exten => chat,1,Goto(100)
> exten => chat,2,MeetMe(${MM},dx1q)
>
> exten => chat,100,MeetMe(${MM},daAx1q)
>
> exten => h,1,MeetMeAdmin(${MM},K)
> +
>
> On Mon, May 9, 2011 at 4:02 PM, Ioan Indreias  wrote:
> > I have tested the following dialplan and it could be used as a
> > starting point. What you have to resolve is how to generate different
> > MeetMe conference room - in the example we have only one room = 1234
> >
> > If you prefix the dialled extension with 1 => you will have a "lovely
> > chat". With 2 -> "cursing chat".
> >
> > HTH,
> >
> > Ioan
>
> --
> _
> -- 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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-- 
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Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.

V: +92 (0)  6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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Re: [asterisk-users] Background music during a call

[Rizwan Hisham]

Will this work:

exten=> 123,1,Meetme(1234)
exten=> 123,n,Hangup()

exten=> 5000,1,Dial(Local/123@bk_music/n,,m())
exten=> 5000,2,Goto(bk_music,123,1)

Parties can call 123 to enter a meeting room. and with the help of a
callfile ic an dial a local channel to 5000 extension which in return calls
a local channel to exten 123 to enter meet me. The dial command with second
local channel will use m() option with moh call defind for each caller. will
ring indefinately with moh and conf members will listen to it.

Not tested it yet. Just sharing, will try it and let you know list.

Cheers

 Mon, May 9, 2011 at 8:47 AM, Rizwan Hisham  wrote:

> Thanks for the reply. I looked into the G option of Dial applications. No
> problem with that but How do I create a ghost call?
>
> My dial plan will look like this:
>
> Caller A calls Caller B normally:
>
> exten=> _XXX,1,SomePreDialApps()
> exten=> _XXX,n,Dial(SIP/B)
> exten=> _XXX,n,Hangup()
>
> Caller A calls caller B ith background music
> exten=> _*9XXX,1,SomePreDialApps()
> exten=> _*9XXX,n,Dial(SIP/B,,G(10))
> exten=> _*9XXX,n,Hangup()
>
> exten=> _*9XXX,10,Goto(mm,1,1)
> exten=> _*9XXX,11,Goto(mm,1,1)
>
> Waiting for your replies
>
> Thanks
>
> On Fri, May 6, 2011 at 10:37 PM, Ioan Indreias  wrote:
>
>> On Fri, May 6, 2011 at 6:30 PM, Rizwan Hisham 
>> wrote:
>>
>> > I am in desperate need of this feature. I want to play background music
>> > during a call while the 2 parties are having some lovely conversation
>> (or
>> > maybe give them a sort of cursing background if they are cursing each
>> > other).
>>
>> Let's start with your actual dialplan (without the background music)
>> and we could start from that point.
>> Hint: I am planning to use option G of the Dial application + a meetme
>> room where a "ghost" call will play the specified MOH class
>> (lovely/cursing).
>>
>> HTH,
>> Ioan.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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
>
>


-- 
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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Re: [asterisk-users] Background music during a call

[Rizwan Hisham]

Thanks for the reply. I looked into the G option of Dial applications. No
problem with that but How do I create a ghost call?

My dial plan will look like this:

Caller A calls Caller B normally:

exten=> _XXX,1,SomePreDialApps()
exten=> _XXX,n,Dial(SIP/B)
exten=> _XXX,n,Hangup()

Caller A calls caller B ith background music
exten=> _*9XXX,1,SomePreDialApps()
exten=> _*9XXX,n,Dial(SIP/B,,G(10))
exten=> _*9XXX,n,Hangup()

exten=> _*9XXX,10,Goto(mm,1,1)
exten=> _*9XXX,11,Goto(mm,1,1)

Waiting for your replies

Thanks

On Fri, May 6, 2011 at 10:37 PM, Ioan Indreias  wrote:

> On Fri, May 6, 2011 at 6:30 PM, Rizwan Hisham 
> wrote:
>
> > I am in desperate need of this feature. I want to play background music
> > during a call while the 2 parties are having some lovely conversation (or
> > maybe give them a sort of cursing background if they are cursing each
> > other).
>
> Let's start with your actual dialplan (without the background music)
> and we could start from that point.
> Hint: I am planning to use option G of the Dial application + a meetme
> room where a "ghost" call will play the specified MOH class
> (lovely/cursing).
>
> HTH,
> Ioan.
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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] Background music during a call

[Rizwan Hisham]

Hi All,
I am in desperate need of this feature. I want to play background music
during a call while the 2 parties are having some lovely conversation (or
maybe give them a sort of cursing background if they are cursing each
other). I found this post which talks about creating a ghost call with the
help of queues and putting that queue in a meetme room where queue will play
the song/curse and the two parties will enjoy/maybe not. I could'nt make it
to work.

Anyways, I think this feature will require some creative dialplanning as its
not supported by default by the software. If anyone can tell me how to
create a ghost call then the rest I may be able to figure out myself. If
there is another way plz share coz im on a deadline.

I am a wee bit of a programmer also, so if your idea needs changes in the
code please dont hesitate to share, otherwise you WILL get a call from me
with a special background noise crafted just for you :)

Meanwhile i'll try my best to come up with a solution.

Cheers

-- 
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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Re: [asterisk-users] odbc error - server is gone

[Rizwan Hisham]

isql?

On Sat, Apr 30, 2011 at 6:18 PM, Pezhman Lali  wrote:

> check your odbc connection with isql
>
> best
>
>
>
> On Fri, Apr 29, 2011 at 9:33 PM, Warren Selby wrote:
>
>> You're using 1.4.2. Why not try upgrading to a more recent release of 1.4
>> (I believe 1.4.41 is current) and see if your issue has been resolved.
>>
>> Thanks,
>> --Warren Selby, dCAP
>>
>> On Apr 29, 2011, at 7:32 AM, Rizwan Hisham 
>> wrote:
>>
>> Yes I have it there, here the content of the file:
>>
>> i think the code is buggy,
>>
>> here is a comment from the function which generated the error
>> (ast_odbc_smart_execute in res_odbc.c line 155 )
>>
>> /* This is a really bad method of trying to correct a dead connection.  It
>>  * only ever really worked with MySQL.  It will not work with any other
>>  * database, since most databases prepare their statements on the server,
>>  * and if you disconnect, you invalidate the statement handle.  Hence, if
>>  * you disconnect, you're going to fail anyway, whether you try to execute
>>  * a second time or not.
>>  */
>>
>> This function is used all over.
>>
>> On Fri, Apr 29, 2011 at 12:23 AM, Sherwood McGowan 
>> <
>> sherwood.mcgo...@gmail.com> wrote:
>>
>>> On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham <
>>> rizwanhas...@gmail.com> wrote:
>>>
>>>> Hi list,
>>>> yesterday I converted my voicemail.conf to realtime voicemail and also
>>>> configured to store the voicemessages in a database using odbc as described
>>>> here <http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail>and
>>>> here<http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage>
>>>> .
>>>> I am using asterisk 1.4.2 with mysql. I also installed the proper odbc
>>>> driver for mysql on the server. I successfully completed the conversion of 
>>>> a
>>>> lot of voicemail users into db yesterday. But today on the CLI thsi error
>>>> was showing;
>>>>
>>>> [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute:
>>>> SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
>>>> Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
>>>> [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute:
>>>> SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
>>>> Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
>>>> [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL
>>>> Execute error!
>>>> [SELECT COUNT(*) FROM voicemessages WHERE dir =
>>>> '/var/spool/asterisk/voicemail/default/1757XXX/INBOX']
>>>>
>>>> I know that the error is caused due to stale odbc connection with mysql.
>>>> But i want to find out if there is a cure for it. Why the connection went
>>>> stale in the first place also.
>>>>
>>>> Any ideas?
>>>>
>>>> --
>>>> Best Ragards
>>>> Rizwan Qureshi
>>>> VoIP/Asterisk Engineer
>>>> Axvoice Inc.
>>>>
>>>> V: +92 (0)  6767 26
>>>> E: rizwanhas...@gmail.com
>>>> W: <http://www.axvoice.com/>www.axvoice.com
>>>>
>>>>
>>>> --
>>>> _
>>>> -- Bandwidth and Colocation Provided by <http://www.api-digital.com>
>>>> http://www.api-digital.com --
>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>   <http://www.asterisk.org/hello>
>>>> http://www.asterisk.org/hello
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>   <http://lists.digium.com/mailman/listinfo/asterisk-users>
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>
>>> do you have "sanitysql => select 1" configured in res_odbc.ini?
>>>
>>> --
>>> Sherwood McGowan
>>> Telecommunications and VOIP Consultant
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by <http://www.api-digital.com>
>>> http://www.api-digital.com --
&g

Re: [asterisk-users] odbc error - server is gone

[Rizwan Hisham]

Yes I have it there, here the content of the file:

i think the code is buggy,

here is a comment from the function which generated the error
(ast_odbc_smart_execute in res_odbc.c line 155 )

/* This is a really bad method of trying to correct a dead connection.  It
 * only ever really worked with MySQL.  It will not work with any other
 * database, since most databases prepare their statements on the server,
 * and if you disconnect, you invalidate the statement handle.  Hence, if
 * you disconnect, you're going to fail anyway, whether you try to execute
 * a second time or not.
 */

This function is used all over.

On Fri, Apr 29, 2011 at 12:23 AM, Sherwood McGowan <
sherwood.mcgo...@gmail.com> wrote:

> On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham wrote:
>
>> Hi list,
>> yesterday I converted my voicemail.conf to realtime voicemail and also
>> configured to store the voicemessages in a database using odbc as described
>> here <http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail> and
>> here <http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage>
>> .
>> I am using asterisk 1.4.2 with mysql. I also installed the proper odbc
>> driver for mysql on the server. I successfully completed the conversion of a
>> lot of voicemail users into db yesterday. But today on the CLI thsi error
>> was showing;
>>
>> [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute:
>> SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
>> Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
>> [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute:
>> SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
>> Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
>> [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL
>> Execute error!
>> [SELECT COUNT(*) FROM voicemessages WHERE dir =
>> '/var/spool/asterisk/voicemail/default/1757XXX/INBOX']
>>
>> I know that the error is caused due to stale odbc connection with mysql.
>> But i want to find out if there is a cure for it. Why the connection went
>> stale in the first place also.
>>
>> Any ideas?
>>
>> --
>> Best Ragards
>> Rizwan Qureshi
>> VoIP/Asterisk Engineer
>> Axvoice Inc.
>>
>> V: +92 (0)  6767 26
>> E: rizwanhas...@gmail.com
>> W: www.axvoice.com
>>
>>
>> --
>> _
>> -- 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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> do you have "sanitysql => select 1" configured in res_odbc.ini?
>
> --
> Sherwood McGowan
> Telecommunications and VOIP Consultant
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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] odbc error - server is gone

[Rizwan Hisham]

Hi list,
yesterday I converted my voicemail.conf to realtime voicemail and also
configured to store the voicemessages in a database using odbc as described
here  and
here .
I am using asterisk 1.4.2 with mysql. I also installed the proper odbc
driver for mysql on the server. I successfully completed the conversion of a
lot of voicemail users into db yesterday. But today on the CLI thsi error
was showing;

[Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL
Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
[Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL
Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
[Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL
Execute error!
[SELECT COUNT(*) FROM voicemessages WHERE dir =
'/var/spool/asterisk/voicemail/default/1757XXX/INBOX']

I know that the error is caused due to stale odbc connection with mysql. But
i want to find out if there is a cure for it. Why the connection went stale
in the first place also.

Any ideas?

-- 
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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Re: [asterisk-users] CDR & ARI Question

[Rizwan Hisham]

google for adaptive cdr. in asterisk.

On Sun, Apr 17, 2011 at 3:58 AM, John Jolly  wrote:

> I have a particular DID that when called will prompt the user to enter the
> caller id that they want to be displayed followed by it prompting for the
> phone number to dial. How would I go about getting thest calls logged in
> both CDR and ARI? Currently, only the callerid information from the original
> caller is populated in CDR and ARI. Is there a way to get this call detail
> to be logged in the CDR and ARI databases?
>
> My current dialplan looks like this -- the calls do get recorded but there
> is no link in ARI to the actual recorded file(s).
>
> exten =>
> s,6,Set(CALLFILENAME=${calleridnum}--${STRFTIME(${EPOCH},,%d-%m-%Y-%H:$
> exten => s,7,MixMonitor(${CALLFILENAME}.wav)
> exten => s,8,Dial(SIP/flowroute/${telnum})
> exten => s,9,StopMonitor()
>
> Any help/advice is greatly appreciated...
>
> J. Jolly
> jgjolly[at]gmail.com
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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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Re: [asterisk-users] call-limit bypass

[Rizwan Hisham]

I am using asterisk 1.4.2 and it usually does enforce the limit. yesterday
and couple of times before was an exception. I am still trying to find the
reason behind. Any more suggestions please?

oh by the way * 1.8.1.1 does enforce the call limit, i tested it yesteday on
sip channels.

On Mon, Apr 4, 2011 at 11:48 PM, Bryant Zimmerman wrote:

> From what I understand on the newer versions of asterisk call-limit does
> not limit calls anymore. You have to limit them from your code using call
> groups.
> From what I have seen on the 1.6x and 1.8 versions call-limit does not
> limit your call counts. We use code and the GROUP_COUNT to limit calls. If
> you use it right it is rock solid.
>
> Thanks
>
> Bryant Zimmerman (ZK Tech Inc.)
> 616-855-1030 Ext. 2003
>
>
> ------
> *From*: "Rizwan Hisham" 
> *Sent*: Monday, April 04, 2011 12:30 PM
> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> *Subject*: [asterisk-users] call-limit bypass
>
>
> Hi everyone,
> one of our users last night bypassed asterisk call-limit limitation. I have
> no Idea how. Is it possible? Is there a bug in asterisk that can be
> manipulated for this purpose?
>
> The call-limit variable was to 2, and the user initiated 169 calls in 2
> minutes each has duration at least 8 minutes.
>
> Please comment...
>
> Thanks
>
> --
>  Best Ragards
> Rizwan Qureshi
> VoIP/Asterisk Engineer
> Axvoice Inc.
>
>  V: +92 (0)  6767 26
> E: rizwanhas...@gmail.com
> W: www.axvoice.com
>
>
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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] call-limit bypass

[Rizwan Hisham]

Hi everyone,
one of our users last night bypassed asterisk call-limit limitation. I have
no Idea how. Is it possible? Is there a bug in asterisk that can be
manipulated for this purpose?

The call-limit variable was to 2, and the user initiated 169 calls in 2
minutes each has duration at least 8 minutes.

Please comment...

Thanks

-- 
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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Re: [asterisk-users] call forwarding

[Rizwan Hisham]

Do this:

exten=> _0522XX,1,Goto(${CONTEXT},0520${EXTEN:4:},1)

you can also use the dial command for this as well

exten=> _0522XX,1,Dial(Local/0520${EXTEN:4}@${SOMECONTEXT})

replace ${CoNTEXT} and ${SOMECONTEXT} with name of your contexts which
contains 0520 numbers.

I have not tested it, you can try it on your setup.


On Mon, Apr 4, 2011 at 7:00 PM, salaheddine elharit <
salah.elharit...@gmail.com> wrote:

> Hello list,
>
> i have one question related to call forwarding.
>
> i have 2 number for the inbound and i want to configure asterisk like that.
>
> When the customer call the first number 0522XX the call will be
> forwarding automatically to anther number 0520xx
>
> Does anybody have a solution to this problem.
>
> Thanks and Regards.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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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Re: [asterisk-users] Fwd: Asterisk 1.6.2.10 & CDR custom added field

[Rizwan Hisham]

You have to use adaptive cdr for this functionality. In 1.8 the conf file
for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf file should tell
you everything.

If you are using some other cdr engine then you will have to jump into the
code of asterisk to make it log the item you want, which includes creating
an extra variable in the cdr data struction, creating a function to set/get
its value from dialplan, and then changing the sql command to include the
extra variable for insertion into DB.

On Thu, Mar 24, 2011 at 1:55 PM, Jonas Kellens wrote:

>  Hello,
>
> is there anyone who can point me to correct information ?
>
> Following http://pbxinaflash.com/forum/showthread.php?t=9042 and
> http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql > Extending CDR does
> not result in a working environment for me.
>
> Any feedback appreciated.
>
>
>
> Kind regards,
> Jonas.
>
>
>
>  Original Message   Subject: [asterisk-users] Asterisk
> 1.6.2.10 & CDR custom added field  Date: Tue, 22 Mar 2011 14:05:23 +0100  
> From:
> Jonas KellensReply-To:
> Asterisk Users Mailing List - Non-Commercial Discussion
>To: 
> Asterisk
> Users Mailing List - Non-Commercial Discussion
>  
>
>
> Hello list,
>
> I have added an extra field "mycolumn" to the cdr table in my MySQL-DB.
>
> I simply try to add a value to this column by doing the following in the
> dialplan :
>
> exten => 600,n,Set(CDR(mycolumn)="myvalue")
>
>
> But this value is not written to the column 'mycolumn' together with the
> other CDR-data.
>
> Why is this ?! Do I need further configuration ? (Not according to
> http://pbxinaflash.com/forum/showthread.php?t=9042)
>
>
> Kind regards,
> Jonas.
>
>
> --
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Re: [asterisk-users] AMI Timestamps unit

[Rizwan Hisham]

Never mind. Its in seconds :)

On Tue, Mar 15, 2011 at 6:48 PM, Rizwan Hisham wrote:

> Hi all,
> What is the unit of asterisk AMI events timestamp value?
>
> milli/micro etc ?
>
> --
> 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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Re: [asterisk-users] Multiple Asterisk

[Rizwan Hisham]

Here is a better link for DUNDi

http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/

skip the part which you know already

On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes wrote:

>
> []'sf.rique
>
>
> On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger wrote:
>
>> On 11-03-15 06:19 PM, Henrique Fernandes wrote:
>>
>>> Have many diferenet locations that have convencional phones that need to
>>> call others locations with convencional phones. And we can not change
>>> this,
>>> I was reading and asterisk cannot handle it self this kind of setup, it
>>> needs an separated serrver to control and routers the calls to this poins
>>> right ?
>>>
>>> So can you guys give any help ? I guess asterisk with SER could do the
>>> job ?
>>>
>>>  I don't believe SER will help you in the setup (see below).
>>
>>
>>  So my question is how do i make the 2 PABX with asterisk talk to  each
>>> other?  Do i need only 2 asterisk with digium or i need one server with
>>> SER
>>> to maki it happen ? There is another program that does what i am looking
>>> for
>>> ?
>>>
>>>  If you require local hardware for each site, then you can install
>> Asterisk at each location.  You can then interconnect them using IAX2 or
>> SIP, additionally you can use DUNDi in your dialplans to share information
>> before the Asterisk boxes.
>>
>
> Thanks!
>
> I had heard some thing about DUNDi but now i am reading i guess it is what
> i need!
>
> I am guessing i can use both IAX2 and SIP i read something about H.323
>
> So i am gonna see which one is best to conect the Asterisk PBX if i am not
> able to use bot SIP and IAX2
>
> Thanks!
>
> here is a link that explains better what DUNDi is!
>
> http://www.voip-info.org/wiki/view/DUNDi
>
>
>> --
>> Paul Belanger
>> Digium, Inc. | Software Developer
>> twitter: pabelanger | IRC: pabelanger (Freenode)
>> Check us out at: http://digium.com & http://asterisk.org
>>
>> --
>> _
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>>
>
>
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Re: [asterisk-users] call being rejected

[Rizwan Hisham]

Isrlgb,
Its the output from "dialplan show" command. The actual entry for the
extension has to be like what you said.

On Tue, Mar 15, 2011 at 7:16 PM,  wrote:

> Shouldn't that be
> Exten =>   1104, 1, Goto(smvoice-mediaport-public-address,s,1)
> -Original Message-
> From: Rizwan Hisham 
> Sender: asterisk-users-boun...@lists.digium.com
> Date: Tue, 15 Mar 2011 19:03:33
> To: Asterisk Users Mailing List - Non-Commercial Discussion<
> asterisk-users@lists.digium.com>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Subject: Re: [asterisk-users] call being rejected
>
> --
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Re: [asterisk-users] call being rejected

[Rizwan Hisham]

You can try changing the priority of

'1104' => 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config]

tp this

'1104' => 2. Goto(smvoice-mediaport-public-address,s,1) [pbx_config]


On Tue, Mar 15, 2011 at 6:59 PM, Jerry Geis  wrote:

> I am using asterisk 1.8.3.
>
> I am getting this error:
> [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite:
> Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because
> extension not found in context 'smvoice-mediaport'.
>
> "dialplan show"  gives me that the context is present:
>
> [ Context 'smvoice-mediaport' created by 'pbx_config' ]
>  '1104' => 1. Goto(smvoice-mediaport-public-address,s,1)
> [pbx_config]
>  'mediaport_direct' => 1. Goto(smvoice-mediaport-public-address,s,1)
> [pbx_config]
>  'public_address' => 1. Goto(smvoice-mediaport-public-address,s,1)
> [pbx_config]
>
>
> the server is showing the call to 1104 which is valid :
>  -- Executing [1104@smvoice-sip:1] Dial("SIP/528-0124",
> "SIP/mndemo_to_vizioconfrm104/1104") in new stack
>  == Using SIP RTP CoS mark 5
>
>
> Why is my call not going through?
>
> Jerry
>
>
>
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[asterisk-users] AMI Timestamps unit

[Rizwan Hisham]

Hi all,
What is the unit of asterisk AMI events timestamp value?

milli/micro etc ?

-- 
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Re: [asterisk-users] Multiple SIP endpoint registrations

[Rizwan Hisham]

You can register multiple end users with only one sip account but asterisk
does not support ringing all the registered phones on single account.
Whenever a new registration comes, asterisk updates its contact info in
memory. So if the registration is coming from multiple end users (multiple
ip address and port) then the call will be placed to the phone who sent
latest registration request. Asterisk does not keep track of all the ip
addresses for single account registration.

What we have done to ring all the end users with same account is that we
listen to registration requests thru manager api in order to detect multiple
registration. If we have detected multiple registration then we store the
contact information of all the end user phones which are related to single
account. And when asterisk receives a dial request for that user, we create
a temporary/fake users (as many as needed) in memory and dial all of them in
the code not thru Dial application as it does not support thsi scenario.

We are still working on this scenario. It is in working condition but in
testing phase.

On Wed, Mar 9, 2011 at 4:14 PM, --[ UxBoD ]--  wrote:

> Hi,
>
> With Asterisk 1.8 is it now possible to register the same SIP account at
> multiple endpoints and for both to ring when the associated extension is
> dialed ?
> --
> Thanks, Phil
>
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Re: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP

[Rizwan Hisham]

1.8 supports static peers along with realtime peers. I have tested.

On Mon, Feb 21, 2011 at 11:04 PM, Ricardo Carvalho <
rjcarvalho.li...@gmail.com> wrote:

> Thanks Faisal, in fact I made a test that confirmed that in realtime
> asterisk doesn’t supported static peers, like you told me.
> Do you know if newer versions of asterisk, like 1.8, have this issue
> already solved?
>
> Regards,
> Ricardo.
>
>
>
>
> On Wed, Feb 16, 2011 at 6:26 PM, Faisal Hanif  wrote:
>
>> I have played a lot on this issue with asterisk config but in realtime it
>> doesn’t supported static peers with version 1.6.2.14.
>>
>>
>>
>> *From:* Ricardo Carvalho [mailto:rjcarvalho.li...@gmail.com]
>> *Sent:* Wednesday, February 16, 2011 10:21 PM
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Cc:* Faisal Hanif
>> *Subject:* Re: [asterisk-users] trunk not working if I register a phone
>> at the same IP as the trunk peer's IP
>>
>>
>>
>> Isn't this a limitation that can be surpassed with some configuration that
>> I'm lacking in my sip.conf or extensions.conf of my asterisk?
>>
>>
>>
>> Ricardo.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif  wrote:
>>
>> Well a quick n easy fix for you is you can configure you call sending
>> peers to use username & secret in INVITE. As far as I know it possible in
>> almost all CISCO, Avaya and all other standard Gateway and SBCs which
>> follows full SIP RFCs.
>>
>>
>>
>> If you can’t do it then you need to use curl as realtime engine instead of
>> MySQL. It will call a URL for each SIP request which you can handle with
>> flexibility in your CGI scripts with apache. But be careful as per my
>> experience asterisk 1.6 with curl as realtime engine can handle a max of 120
>> registration in parallel if registration refresh time is 120 seconds.
>>
>>
>>
>> Faisal Hanif
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ricardo Carvalho
>> *Sent:* Wednesday, February 16, 2011 9:41 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] trunk not working if I register a phone at
>> the same IP as the trunk peer's IP
>>
>>
>>
>> How should I configure my asterisk server so that I can receive calls from
>> an unregistered peer from whom I also receive registrations of sip phones?
>>
>>
>>
>> I'm asking you this, because with my actual configuration, when I register
>> a contact from that peer's IP, no more inbound calls are accepted from that
>> peer, as my asterisk rejects those INVITEs with "407 Proxy Authentication
>> Required", I assume because they don't carry the registered contact
>> registration!!!
>>
>> My SIP contacts have type=friend and all inbound calls not coming from my
>> registered phones fall in the default context without authentication, so
>> that someone in the Internet be able to call freely through the Internet
>> anyone in my server's dial plan.
>>
>>
>>
>> Some ideas?
>>
>>
>>
>> Regards,
>>
>> Ricardo Carvalho.
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
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>>
>>
>>
>
>
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Re: [asterisk-users] (fast) AGI and AMI synchronization ?

[Rizwan Hisham]

You can use threads and queues in your program to interface with AMI. Your
main thread should get all the events from * and based on some logic enqueue
it. Some other thread should be listening to the queue and in that thread
you are free to read the input whenever you want. This way you are free to
keep listening from the AMI as fast as you can and also your other thread
will process them at its own pace.

Cheers

On Tue, Mar 8, 2011 at 5:27 PM, Faisal Hanif  wrote:

> AMI is single threaded link so waiting on it will bring things to hang mode
> but FastAGI & dialplan is multithread. Better to manage all info by AMI in
> a
> local hash or array and use sleep/waiting on AGI till required info
> populated to hash/array by AMI.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Corentin Le
> Gall
> Sent: Tuesday, March 08, 2011 4:31 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] (fast) AGI and AMI synchronization ?
>
> Hi,
>
> I've been developing some CTI software around asterisk for a while, mainly
> with the help of AMI and fast AGI.
> It works quite fine, but I have some trouble sometimes with the
> un-synchronized property of these 2.
> Let me explain, we have a dialplan like this one :
>
> exten = s,n,UserEvent(useful_input_data)
> (...) a few actions
> exten = s,n,AGI(agi://127.0.0.1:/fetch,queuename)
>
> The idea is to setup a "cti server" that talks with both AMI and AGI
> channels, the first one mainly when one just want to send some data from
> asterisk to the "cti server", and the second one when the dialplan needs
> some data from this server.
>
> My issue is that the AGI requests are received (from the CTI server point
> of
> vue) a little bit before the AMI events. In most cases, I don't really care
> because it is only "a little", and the data asterisk needs to fetch from
> the
> AGI are set on time. But sometimes not, especially in cases like above,
> when
> there are only a few dialplan lines between UserEvent and AGI ...
>
> In order to handle that, I thought "let's make a sync/meeting point, with
> the help of the AMI NewExten event, when the app is AGI".
> The idea would be to keep the AGI connection open as long as the good AMI
> NewExten event is not received, then to reply and close it, in order for
> the
> dialplan to proceed.
> However, when trying to do this, nothing more occurs on the AMI connection,
> thus I come to a deadlock ...
>
> My question is then, before switching to -dev issues : is there an option
> somewhere to handle this, whether on the AMI or on the AGI side ?
> The asterisk version we've been using for a long time is 1.4 and my current
> attempts are done on 1.8 branch.
>
> Thanks,
> --
> Corentin LE GALL
> Proformatique (Groupe Avencall) - 10bis rue Lucien Voilin - F-92800 Puteaux
> Tel (+33/0)1.41.38.99.60 - Fax (+33/0)1.41.38.99.70 http://wiki.xivo.fr
>
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Re: [asterisk-users] [asterisk-dev] CDR and call transfers :)

[Rizwan Hisham]

Anymore suggestions please.

On Tue, Mar 8, 2011 at 3:36 PM, Rizwan Hisham wrote:

> Thanks Klaus,
> Actually I got my idea from CEL. But I am more familiar with AMI, plus CEL
> generates too many events for a single call. I dont want that, I already
> have a library of routines which read manager events, i just have to plug in
> my new idea. But still I will think about CEL once again. I never gave it a
> second thought.
>
> I like your idea of a gateway asterisk as well. Will try it.
>
> Thanks
>
>
> On Tue, Mar 8, 2011 at 3:27 PM, Klaus Darilion <
> klaus.mailingli...@pernau.at> wrote:
>
>>
>>
>> Am 08.03.2011 11:05, schrieb Rizwan Hisham:
>> > Hi all,
>> > I have a problem with CDRs when doing call transfers. I am using *
>> 1.8.2.3
>> > with cdr_odbc.
>>
>> > This is the best supposed solution i have come up with. But, I am here
>> to
>> > ask you people for your ideas and thoughts on my solution. I am still in
>> > search for a better solution. So please share your ideas.
>>
>> Sounds like you are trying to re-implement CEL:
>>
>>
>> https://wiki.asterisk.org/wiki/display/AST/Call+Event+Log+%28CEL%29+Driver+Modules
>> https://wiki.asterisk.org/wiki/display/AST/CEL+Design+Goals
>>
>> klaus
>>
>> PS: I prefer a dedicated GW-Asterisk which does accounting:
>>
>>   /---trunk1
>>  /
>> /-trunk2
>>  SIP > PBX  <> GW Asterisk <---ISDN
>> phones   Asterisk   \--ISDN2
>> \---...
>>
>>
>> So, all the transfers happens in the PBX Asterisk. All calls which will
>> be billed are routed via the GW-Asterisk into the PSTN via several
>> uplinks or back to the same or another PBX Asterisk. So, I generate CDRs
>> only at the GW-Asterisk, and as there never happens any transfers on the
>> GW-Asterisk, those CDRs are always 100% correct (as long as you signal
>> proper CLIs from the PBX Asterisk to the GW-Asterisk).
>>
>>
>
>
> --
> 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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Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0)  6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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Re: [asterisk-users] [asterisk-dev] CDR and call transfers :)

[Rizwan Hisham]

Thanks Klaus,
Actually I got my idea from CEL. But I am more familiar with AMI, plus CEL
generates too many events for a single call. I dont want that, I already
have a library of routines which read manager events, i just have to plug in
my new idea. But still I will think about CEL once again. I never gave it a
second thought.

I like your idea of a gateway asterisk as well. Will try it.

Thanks

On Tue, Mar 8, 2011 at 3:27 PM, Klaus Darilion  wrote:

>
>
> Am 08.03.2011 11:05, schrieb Rizwan Hisham:
> > Hi all,
> > I have a problem with CDRs when doing call transfers. I am using *
> 1.8.2.3
> > with cdr_odbc.
>
> > This is the best supposed solution i have come up with. But, I am here to
> > ask you people for your ideas and thoughts on my solution. I am still in
> > search for a better solution. So please share your ideas.
>
> Sounds like you are trying to re-implement CEL:
>
>
> https://wiki.asterisk.org/wiki/display/AST/Call+Event+Log+%28CEL%29+Driver+Modules
> https://wiki.asterisk.org/wiki/display/AST/CEL+Design+Goals
>
> klaus
>
> PS: I prefer a dedicated GW-Asterisk which does accounting:
>
>   /---trunk1
>  /
> /-trunk2
>  SIP > PBX  <> GW Asterisk <---ISDN
> phones   Asterisk   \--ISDN2
> \---...
>
>
> So, all the transfers happens in the PBX Asterisk. All calls which will
> be billed are routed via the GW-Asterisk into the PSTN via several
> uplinks or back to the same or another PBX Asterisk. So, I generate CDRs
> only at the GW-Asterisk, and as there never happens any transfers on the
> GW-Asterisk, those CDRs are always 100% correct (as long as you signal
> proper CLIs from the PBX Asterisk to the GW-Asterisk).
>
>


-- 
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] CDR and call transfers :)

[Rizwan Hisham]

Hi all,
I have a problem with CDRs when doing call transfers. I am using * 1.8.2.3
with cdr_odbc.

As most of you may already know, CDRs and call transfers dont go along very
well in *. I mean the developers team have done their best to bring it to an
acceptable level. But still it cannot meet the needs of all of the users,
some like me are too

Whats happening here is when i transfer a call upto two or three or more
times, not all the cdrs are generated all the time, hence if i lose one of
them, i lose money. It happens with both attended and blind transfers. Bad
news is my system cannot afford to not have call transfers facility. I also
created call transfers in dialplan with the combination of call parkling and
blind transfer (blind transfer seems to generate correct cdrs most of the
times). Anyways it did not work (call transfer worked but CDRs didnt)

Now I am working on another plan. I am using the builtin transfer facility
of * but I have modified some of the code of these features so that whenever
these features execute, they send a manager event stating a transfer occured
with the following information:

Event: Dial
Privilege: call,all
Timestamp: 1299577784.825096
SubEvent: *Blind Transfer*
*Transferer: SIP/pepsi-0002
Transferee: SIP/coke-0003*
UniqueID1: 1299577741.2
UniqueID2: 1299577741.3
LinkedID1: 1299577741.2
LinkedID2: 1299577741.3
Transfer To: 17142545586
Transfer Context: siga-external

I plan to watch for this event thru AMI, and record who was invloved in
transfers, hopefully correct the bill sec and duration with the help of some
other events and their timestamps (UnLink Event, Hangup Event etc) in the
cdr after it has been inserted in DB, or if its not there in DB, i will
insert my own :)

This is the best supposed solution i have come up with. But, I am here to
ask you people for your ideas and thoughts on my solution. I am still in
search for a better solution. So please share your ideas.

Thanks

PS. I am sending this message to both users and developers list coz i am not
sure where this message truly belongs.

-- 
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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Re: [asterisk-users] 2 ip phones and 1 normal, can't neither send nor receive calls at all...

[Rizwan Hisham]

If you can post your extensions.conf, sip.conf and features.conf then maybe
some one can understand and help with your problem.

Thanks

On Sat, Mar 5, 2011 at 5:42 AM, Francisco Javier Cintrón Olguín <
fcintr...@gmail.com> wrote:

> I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco
> spa8800, all them are internal lines.
>
> 1.- spa921, 401 ext
> 2.- spa921, 402 ext
> 3.- normal phone connected to spa8800 404 ext.
>
> It had a very strange behavior when I was configuring call transfer and
> call pickup.
>
> These are steps to repeat it:
>
>
> 1.- from 401 call to 404
> 2.- from 404 don't answer it.
> 3.- from 402 press *8 and wait 10 seconds
> 4.- 402 says that it is connected.
> 5.- 404 stops to sound.
> 6.- 401 keeps ringing
> 7.- Hang up 402
> 8.- Hang up 401
>
>
> After these steps I can not neither send nor receive calls from anyone of
> 401, 402 or 404 until I restart asterisk.
>
> /var/log/asterisk/messages, doesn´t show anything strange.
>
> ¿what's happening with my phones?
>
> Thank you for your kind help.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0)  6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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Re: [asterisk-users] Prepaid Billing other than A2Billing

[Rizwan Hisham]

>If you're in the market for a custom solution for whatever reason, there's
more than a few of us who can write a custom prepaid solution. I've done
about 7 so far personally and I know there's more like me out there

Yes you are right. I am now one of them (i took the red pill :)

On Sun, Mar 6, 2011 at 8:39 PM, Sherwood McGowan  wrote:

> If you're in the market for a custom solution for whatever reason, there's
> more than a few of us who can write a custom prepaid solution. I've done
> about 7 so far personally and I know there's more like me out there
>
>
> On Sat, Mar 5, 2011 at 11:26 AM, bilal ghayyad wrote:
>
>> Hi All;
>>
>> Any one advise for open source prepaid billing other than A2Billing that
>> can work with Asterisk and it is rich by features (for large business)?
>>
>> Regards
>> Bilal
>>
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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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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Re: [asterisk-users] Failover Routing

[Rizwan Hisham]

You can use exten pattern matching for un allocated numbers, say exten=>
_X.,1,Goto(somewhere) will match all the numbers on priority 1. But make
sure you match full extension numbers first which are allocated. Also this
extension is a security risk as well. It is recommended that you use a
filter dialplan application/function before matching this extension to make
sure you accept numbers only.

Cheers

On Tue, Mar 1, 2011 at 5:11 PM, Deepika Nijhawan <
deepika.nijha...@oxygen8.com> wrote:

> Hi,
>
>
>
> If I use dialstatus variable, it doesn’t give exact reasons for failure
> like for unallocated numbers it sends Congestion. Whereas, for unallocated
> number I don’t want to go to failover routing. But need to go to failover
> routing for other congestion reasons.
>
> So, is there any way to check SIP responses like 4xx or 5xx ?
>
>
>
> Thanks,
>
> Deepika
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



-- 
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Rizwan Qureshi
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Axvoice Inc.
V: +92 (0)  6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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Re: [asterisk-users] asterisk security....again

[Rizwan Hisham]

Thanks all. I appreciate your support a lot. Your suggestions helped make a
game plan finally. Otherwise i was just shooting in the dark before. Anyways
here is what I am going to do now.

install a better firewall, possibly the one mentioned by mr j stapleton and
tighten its security settings. use IP tables in conjunction with the
firewall. use something like
this<http://etel.wiki.oreilly.com/wiki/index.php/Asterisk_Brute_Force_Prevention>to
prevent bruteforce attacks. making changes in asterisk sip peer
setting
to make it as tight as I can security wise without affecting the service we
provide. Provisioning the remote end user ata to make it accept sip messages
coming from only the proxy (i'll se what else i can do with them). and keep
looking for an even better solution to this problem.

Thanks a lot again.

Cheers

On Mon, Feb 28, 2011 at 11:16 PM, satish patel wrote:

>  It could be possible they are not scanning your asterisk server. They are
> just scanning 5060 and in this case your ATA caught by scan directly that
> why you don't have any logs on server side. Don't you have any setting in
> ATA to specify allowed IP address ?
>
> -Satish
>
> --
> From: jstaple...@computer-business.com
> To: asterisk-users@lists.digium.com
> Date: Mon, 28 Feb 2011 10:27:33 -0500
>
> Subject: Re: [asterisk-users] asterisk securityagain
>
> http://sipera.com/ is one such product.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham
> *Sent:* Monday, February 28, 2011 9:33 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] asterisk securityagain
>
>
>
> Thanks Mr. Kevin.
>
> Can anyone please also tell me which firewall is best suited for
> asterisk/sip attack prevention. Is there any firewall built specially to
> address sip security problems?
>
> On Mon, Feb 28, 2011 at 6:38 PM, Kevin P. Fleming 
> wrote:
>
> On 02/28/2011 07:27 AM, Rizwan Hisham wrote:
>
> Any suggestions on encrypting the sip and rtp. I have done some googling
> on it. looks like it is not supported by most end point devices or
> service providers. But still your thoughts will be appreciated on this
> subject.
>
>
>
> You cannot protect a remote SIP endpoint from attacks via your server; that
> SIP endpoint is an endpoint itself, and if it can receive IP packets from
> attackers, it will process them. These packets don't go through your server,
> and encrypting the legitimate traffic between your server and the remote
> endpoint isn't going to make any difference at all.
>
> The *only* way to address attacks like this is to modify the configuration
> of the remote endpoint to ignore all incoming packets that aren't from your
> server(s). Even that is not a perfect solution, though, because the attacker
> (if they are actually aware of your server and customers) can spoof the IP
> addresses of your server(s) in order to get the remote endpoints to at least
> accept an INVITE (they can't place a successful call through them using
> spoofing though).
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
>
>
>
> --
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
>
> 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? Join us for a live introductory webinar every Thurs:
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> or update options visit:
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>
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>   http://www.asterisk.org/hello
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> asterisk-users mai

Re: [asterisk-users] asterisk security....again

[Rizwan Hisham]

Thanks Mr. Kevin.

Can anyone please also tell me which firewall is best suited for
asterisk/sip attack prevention. Is there any firewall built specially to
address sip security problems?

On Mon, Feb 28, 2011 at 6:38 PM, Kevin P. Fleming wrote:

> On 02/28/2011 07:27 AM, Rizwan Hisham wrote:
>
>> Any suggestions on encrypting the sip and rtp. I have done some googling
>> on it. looks like it is not supported by most end point devices or
>> service providers. But still your thoughts will be appreciated on this
>> subject.
>>
>
> You cannot protect a remote SIP endpoint from attacks via your server; that
> SIP endpoint is an endpoint itself, and if it can receive IP packets from
> attackers, it will process them. These packets don't go through your server,
> and encrypting the legitimate traffic between your server and the remote
> endpoint isn't going to make any difference at all.
>
> The *only* way to address attacks like this is to modify the configuration
> of the remote endpoint to ignore all incoming packets that aren't from your
> server(s). Even that is not a perfect solution, though, because the attacker
> (if they are actually aware of your server and customers) can spoof the IP
> addresses of your server(s) in order to get the remote endpoints to at least
> accept an INVITE (they can't place a successful call through them using
> spoofing though).
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
>
>
> --
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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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Re: [asterisk-users] asterisk security....again

[Rizwan Hisham]

Any suggestions on encrypting the sip and rtp. I have done some googling on
it. looks like it is not supported by most end point devices or service
providers. But still your thoughts will be appreciated on this subject.

On Mon, Feb 28, 2011 at 6:13 PM, Rizwan Hisham wrote:

> You are right Terry. Sorry i did not describe full scenario before. Yes the
> users are remote with atas on port 5060. Attacks on the remote customers was
> my second guess. My network/system admin has already ruled out the
> implementation of VPN. In summary, we dont want to do anything on remote
> customer side. All kind of security and attck prevention techniques have to
> be implemented on the server.
>
> Its comforting to hear someone say "they are harmless". But still i would
> like to know their next step of attack after "guessing/scanning". Or is it
> the only step?
>
> On Mon, Feb 28, 2011 at 5:32 PM, Terry Brummell wrote:
>
>> When he says “customers” I am assuming he means remote customers.  It
>> sounds like he is a reseller of telecom facilities to me.  Which means his
>> customers most likely have ATA’s with port 5060 forwarded to the ATA, or
>> they are direct on the I’net.
>>
>> He has already set the ATA to only allow calls from the proxy server, so
>> sounds like he has plugged the hole.
>>
>>
>>
>> They are not ‘sniffing’ your traffic, they are guessing/scanning.  That’s
>> it, that’s all, no great conspiracy going on.  They look for open 5060, then
>> send SIP requests to it hopefully finding a badly implemented SIP solution
>> to which they can dial through.  Once they determine they cannot get
>> through, the script will move on to the next sucker.
>>
>>
>>
>> You have a couple of options, which you could implement at **each** of
>> your customers if you wanted.  Set up a VPN, tunnel the SIP/RTP traffic
>> through it.  Set up IPTables at the customer to only allow SIP from your
>> IP.  Or, do what you have already done and forget about these idiots doing
>> the scan, they are harmless at this point.
>>
>>
>>
>> Vlans and DMZ for the server do no good as the attacks are being directed
>> at the remote client side, not the server.
>>
>>
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ricardo Carvalho
>> *Sent:* Monday, February 28, 2011 6:31 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] asterisk securityagain
>>
>>
>>
>> Probably, you are receiving INVITE attacks from some tool like sipvicious.
>> You should rearange your network to cover some inportant security issues.
>>
>>
>>
>> The IP address of you server can be revealed in some unincrypted SIP
>> signaling of some call through the Internet to/from your server's client, or
>> simply by your client SRV record in the DNS, if you added it to his DNS.
>>
>>
>>
>> Probably your network is exposed to the Internet. To address those
>> situations, you can use a distinct VLAN to address SIP phones and you also
>> can use port security at the switching ports where you connect your ATAs and
>> phones. You should also deliver with tagging (802.1Q) that VLAN to those
>> ATAs and phones. This should protect you from inside sniffers.
>>
>> This VLAN should just communicate with the DMZ where you should have your
>> asterisk server and between those two networks you should only open the
>> needed ports - for a common SIP infrastructure you should open UDP 5060 and
>> the specified UDP range shown in rtp.conf file for the media to pass. Phones
>> VLAN should not communicate directlly with the world, just in the outbound
>> direction if you like.
>>
>>
>>
>> Regards,
>>
>> Ricardo Carvalho.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Mon, Feb 28, 2011 at 10:33 AM, Rizwan Hisham 
>> wrote:
>>
>> Hi all,
>> The problem I have been experiencing since last month is that some of my
>> customers are getting calls with "Asterisk " caller id. Most of
>> them in the middle of the night. And my asterisk server has no record of
>> these calls. The customers were getting irritated as you can imagine. I
>> guessed the only way to receive incoming calls by by-passing the
>> registration server is thru sip-uri calls directly to customers. I have
>> updated the customers atas to

Re: [asterisk-users] asterisk security....again

[Rizwan Hisham]

You are right Terry. Sorry i did not describe full scenario before. Yes the
users are remote with atas on port 5060. Attacks on the remote customers was
my second guess. My network/system admin has already ruled out the
implementation of VPN. In summary, we dont want to do anything on remote
customer side. All kind of security and attck prevention techniques have to
be implemented on the server.

Its comforting to hear someone say "they are harmless". But still i would
like to know their next step of attack after "guessing/scanning". Or is it
the only step?

On Mon, Feb 28, 2011 at 5:32 PM, Terry Brummell  wrote:

> When he says “customers” I am assuming he means remote customers.  It
> sounds like he is a reseller of telecom facilities to me.  Which means his
> customers most likely have ATA’s with port 5060 forwarded to the ATA, or
> they are direct on the I’net.
>
> He has already set the ATA to only allow calls from the proxy server, so
> sounds like he has plugged the hole.
>
>
>
> They are not ‘sniffing’ your traffic, they are guessing/scanning.  That’s
> it, that’s all, no great conspiracy going on.  They look for open 5060, then
> send SIP requests to it hopefully finding a badly implemented SIP solution
> to which they can dial through.  Once they determine they cannot get
> through, the script will move on to the next sucker.
>
>
>
> You have a couple of options, which you could implement at **each** of
> your customers if you wanted.  Set up a VPN, tunnel the SIP/RTP traffic
> through it.  Set up IPTables at the customer to only allow SIP from your
> IP.  Or, do what you have already done and forget about these idiots doing
> the scan, they are harmless at this point.
>
>
>
> Vlans and DMZ for the server do no good as the attacks are being directed
> at the remote client side, not the server.
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ricardo Carvalho
> *Sent:* Monday, February 28, 2011 6:31 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] asterisk securityagain
>
>
>
> Probably, you are receiving INVITE attacks from some tool like sipvicious.
> You should rearange your network to cover some inportant security issues.
>
>
>
> The IP address of you server can be revealed in some unincrypted SIP
> signaling of some call through the Internet to/from your server's client, or
> simply by your client SRV record in the DNS, if you added it to his DNS.
>
>
>
> Probably your network is exposed to the Internet. To address those
> situations, you can use a distinct VLAN to address SIP phones and you also
> can use port security at the switching ports where you connect your ATAs and
> phones. You should also deliver with tagging (802.1Q) that VLAN to those
> ATAs and phones. This should protect you from inside sniffers.
>
> This VLAN should just communicate with the DMZ where you should have your
> asterisk server and between those two networks you should only open the
> needed ports - for a common SIP infrastructure you should open UDP 5060 and
> the specified UDP range shown in rtp.conf file for the media to pass. Phones
> VLAN should not communicate directlly with the world, just in the outbound
> direction if you like.
>
>
>
> Regards,
>
> Ricardo Carvalho.
>
>
>
>
>
>
>
>
>
>
>
> On Mon, Feb 28, 2011 at 10:33 AM, Rizwan Hisham 
> wrote:
>
> Hi all,
> The problem I have been experiencing since last month is that some of my
> customers are getting calls with "Asterisk " caller id. Most of
> them in the middle of the night. And my asterisk server has no record of
> these calls. The customers were getting irritated as you can imagine. I
> guessed the only way to receive incoming calls by by-passing the
> registration server is thru sip-uri calls directly to customers. I have
> updated the customers atas to not accept any calls from sources other than
> the registration server. Thats all fine now. But the question is how can
> anyone know the direct sip uri addresses of our customers.
>
> My guess is that someone has been sniffing my server's sip traffic. In that
> case what should i do to get rid of the sniffers?
>
> If you think there is another reason for that then please tell me even if
> you dont have the solution.
>
> Thanks
>
> --
>
> Best Ragards
>
> Rizwan Qureshi
>
> VoIP/Asterisk Engineer
>
> Axvoice Inc.
>
> V: +92 (0)  6767 26
>
> E: rizwanhas...@gmail.com
>
> W: www.axvoice.com
>
>
>
>
>
> --
> __

Re: [asterisk-users] asterisk security....again

[Rizwan Hisham]

thanks for the replies.

I dont want to rule-out the possibility of network sniffing. I am sure its
not an inside job. The server is off-site and is hosted by a very well
reputed hosting company. So if someone is sniffing, what should I do?


>Probably, you are receiving INVITE attacks from some tool like sipvicious.
You should rearange your network to cover some inportant security issues.

I have tested sipvicious against my asterisk server already, its been
secured that way.


>Probably your network is exposed to the Internet. To address those
situations, you can use a distinct VLAN to address SIP phones >and you also
can use port security at the switching ports where you connect your ATAs and
phones. You should also deliver with >tagging (802.1Q) that VLAN to those
ATAs and phones. This should protect you from inside sniffers.
>This VLAN should just communicate with the DMZ where you should have your
asterisk server and between those two networks >you should only open the
needed ports - for a common SIP infrastructure you should open UDP 5060 and
the specified UDP >range shown in rtp.conf file for the media to pass.
Phones VLAN should not communicate directlly with the world, just in the
>outbound direction if you like.

I will talk to my network admin about this.

I dont have any wireless network interface to our server. And I am going to
apply that IP table thing to the server.

Any more suggestions please?

On Mon, Feb 28, 2011 at 4:31 PM, Ricardo Carvalho <
rjcarvalho.li...@gmail.com> wrote:

> Probably, you are receiving INVITE attacks from some tool like sipvicious.
> You should rearange your network to cover some inportant security issues.
>
> The IP address of you server can be revealed in some unincrypted SIP
> signaling of some call through the Internet to/from your server's client, or
> simply by your client SRV record in the DNS, if you added it to his DNS.
>
> Probably your network is exposed to the Internet. To address those
> situations, you can use a distinct VLAN to address SIP phones and you also
> can use port security at the switching ports where you connect your ATAs and
> phones. You should also deliver with tagging (802.1Q) that VLAN to those
> ATAs and phones. This should protect you from inside sniffers.
> This VLAN should just communicate with the DMZ where you should have your
> asterisk server and between those two networks you should only open the
> needed ports - for a common SIP infrastructure you should open UDP 5060 and
> the specified UDP range shown in rtp.conf file for the media to pass. Phones
> VLAN should not communicate directlly with the world, just in the outbound
> direction if you like.
>
> Regards,
> Ricardo Carvalho.
>
>
>
>
>
>
> On Mon, Feb 28, 2011 at 10:33 AM, Rizwan Hisham wrote:
>
>> Hi all,
>> The problem I have been experiencing since last month is that some of my
>> customers are getting calls with "Asterisk " caller id. Most of
>> them in the middle of the night. And my asterisk server has no record of
>> these calls. The customers were getting irritated as you can imagine. I
>> guessed the only way to receive incoming calls by by-passing the
>> registration server is thru sip-uri calls directly to customers. I have
>> updated the customers atas to not accept any calls from sources other than
>> the registration server. Thats all fine now. But the question is how can
>> anyone know the direct sip uri addresses of our customers.
>>
>> My guess is that someone has been sniffing my server's sip traffic. In
>> that case what should i do to get rid of the sniffers?
>>
>> If you think there is another reason for that then please tell me even if
>> you dont have the solution.
>>
>> Thanks
>>
>> --
>> Best Ragards
>> Rizwan Qureshi
>> VoIP/Asterisk Engineer
>> Axvoice Inc.
>> V: +92 (0)  6767 26
>> E: rizwanhas...@gmail.com
>> W: www.axvoice.com
>>
>>
>> --
>> _
>> -- 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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>


-- 
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] asterisk security....again

[Rizwan Hisham]

Hi all,
The problem I have been experiencing since last month is that some of my
customers are getting calls with "Asterisk " caller id. Most of
them in the middle of the night. And my asterisk server has no record of
these calls. The customers were getting irritated as you can imagine. I
guessed the only way to receive incoming calls by by-passing the
registration server is thru sip-uri calls directly to customers. I have
updated the customers atas to not accept any calls from sources other than
the registration server. Thats all fine now. But the question is how can
anyone know the direct sip uri addresses of our customers.

My guess is that someone has been sniffing my server's sip traffic. In that
case what should i do to get rid of the sniffers?

If you think there is another reason for that then please tell me even if
you dont have the solution.

Thanks

-- 
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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Re: [asterisk-users] Carrying context from one server to another?

[Rizwan Hisham]

you can also set some kind of authentication on the extensions for example
ask for a pin to dialout. etc

On Thu, Feb 24, 2011 at 6:51 PM, Daniel Tryba  wrote:

> On Thu, Feb 24, 2011 at 11:38:17AM +, Roger Burton West wrote:
> > The relevant part of my setup is something like:
> >
> > SIP phones -> local server -> remote server -> SIP-to-PSTN provider
> >
> > I want _some_ of the SIP phones on the local server to be able to get
> > access to SIP-to-PSTN, but not all of them. The local-to-remote
> > connection is IAX2 over VPN.
>
> The way I would to this is by blocking them on the localserver (with
> different contexts). An other solution would be to set prefixes on the
> extension when dialing from local to remote and use these to filter, not
> very elegant but works over any transport. I use this to do multitenant
> billing on the remote server in places where I only want 1 IAX trunk.
> Whether this is effective depends on your control of the local server.
>
> --
>
>   Daniel Tryba
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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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Re: [asterisk-users] Unknown calls

[Rizwan Hisham]

Its a pure VoIP setup. no cards.

On Thu, Feb 24, 2011 at 7:12 PM, Satish Patel  wrote:

> Do you have PRI card or FXO card?
>
> --
> Sent from my iPhone
>
> On Feb 24, 2011, at 5:28 AM, Rizwan Hisham  wrote:
>
> Thats what im unsure about. I think the calls maybe going to the user
> directly through sip uri or some other method. How can i test that. I have
> already tried to call those customers with direct sip uri dial but does not
> work.
>
> On Thu, Feb 24, 2011 at 3:19 PM, Roger Burton West < 
> ro...@firedrake.org> wrote:
>
>> On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote:
>>
>> >Still last night there was a call to a customer. Plz help me figure out
>> the
>> >solution for this problem.
>>
>> Can you be sure that the call _is_ coming through your Asterisk server,
>> rather than being the result of random scanning for your customers'
>> phones?
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by <http://www.api-digital.com>
>> http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   <http://www.asterisk.org/hello>
>> http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>   <http://lists.digium.com/mailman/listinfo/asterisk-users>
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Best Ragards
> Rizwan Qureshi
> VoIP/Asterisk Engineer
> Axvoice Inc.
> V: +92 (0)  6767 26
> E: rizwanhas...@gmail.com
> W: <http://www.axvoice.com/>www.axvoice.com
>
> --
> _
> -- 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>
> http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
><http://lists.digium.com/mailman/listinfo/asterisk-users>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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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Re: [asterisk-users] Problem in dialing out

[Rizwan Hisham]

try this

http://www.voip-info.org/wiki/view/Asterisk+sip+qualify

On Sat, Feb 19, 2011 at 5:00 AM, asterisk asterisk wrote:

> I have a sip trunk connecting to a huawei softx3000. At the moment, I can
> register and dial in.
>
> However, peer status shows not reachable
>
> sip show peer as follow
>
>   * Name   : cmphone
>   Secret   : 
>   MD5Secret: 
>   Remote Secret: 
>   Context  : from-cmphone
>   Subscr.Cont. : device-hints
>   Language :
>   AMA flags: Unknown
>   Transfer mode: open
>   CallingPres  : Presentation Allowed, Not Screened
>   Callgroup:
>   Pickupgroup  :
>   MOH Suggest  :
>   Mailbox  :
>   VM Extension : asterisk
>   LastMsgsSent : 32767/65535
>   Call limit   : 0
>   Max forwards : 0
>   Dynamic  : No
>   Callerid : "" <>
>   MaxCallBR: 384 kbps
>   Expire   : -1
>   Insecure : port,invite
>   Force rport  : Yes
>   ACL  : No
>   DirectMedACL : No
>   T.38 support : No
>   T.38 EC mode : Unknown
>   T.38 MaxDtgrm: -1
>   DirectMedia  : Yes
>   PromiscRedir : No
>   User=Phone   : No
>   Video Support: No
>   Text Support : No
>   Ign SDP ver  : No
>   Trust RPID   : No
>   Send RPID: No
>   Subscriptions: Yes
>   Overlap dial : Yes
>   Outb. proxy  : 202.0.179.3
>   DTMFmode : rfc2833
>   Timer T1 : 500
>   Timer B  : 32000
>   ToHost   : 202.0.179.3
>   Addr->IP : 202.0.179.3:5060
>   Defaddr->IP  : (null)
>   Prim.Transp. : UDP
>   Allowed.Trsp : UDP
>   Def. Username: 852350xx
>   SIP Options  : 100rel
>   Codecs   : 0xe (gsm|ulaw|alaw)
>   Codec Order  : (alaw:20,ulaw:20,gsm:20)
>   Auto-Framing :  No
>   100 on REG   : No
>   Status   : UNREACHABLE
>   Useragent:
>   Reg. Contact :
>   Qualify Freq : 6 ms
>   Sess-Timers  : Accept
>   Sess-Refresh : uas
>   Sess-Expires : 1800 secs
>   Min-Sess : 90 secs
>   RTP Engine   : asterisk
>   Parkinglot   :
>   Use Reason   : No
>   Encryption   : No
>
> In sip.conf
>
> I have
>
> register = 852350x:secret@202.0.179.3
>
> [cmphone]
> type = friend
> host = 202.0.179.3
> secret = secret
> username = 852350x
> context = from-cmphone
> dtmfmode = rfc2833
> outboundproxy = 202.0.179.3
> caninvite=no
> insecure = port,invite
> nat = yes
>
> When debug is on, the error message is
>
>
> <--- SIP read from UDP:202.0.179.3:5060 --->
> SIP/2.0 504 Server Time-out
> From: "asterisk" ;tag=as2d14b9ec
> To: ;tag=6b0704d0
> CSeq: 102 OPTIONS
> Call-ID: 17e0315c21d7dbc10e8c185740e21...@sip.x.xxx
> Via: SIP/2.0/UDP
> 14.xxx.xxx.xxx:5060;branch=z9hG4bK3646eaf2;received=14.xxx.xxx.xxx;rport=5060
> Content-Length: 0
>
> Any help is appreciate.
>
> CK
>
> --
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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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Re: [asterisk-users] Unknown calls

[Rizwan Hisham]

Thats what im unsure about. I think the calls maybe going to the user
directly through sip uri or some other method. How can i test that. I have
already tried to call those customers with direct sip uri dial but does not
work.

On Thu, Feb 24, 2011 at 3:19 PM, Roger Burton West wrote:

> On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote:
>
> >Still last night there was a call to a customer. Plz help me figure out
> the
> >solution for this problem.
>
> Can you be sure that the call _is_ coming through your Asterisk server,
> rather than being the result of random scanning for your customers'
> phones?
>
>
> --
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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

[Rizwan Hisham]

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  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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-- 
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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] Unknown calls

[Rizwan Hisham]

Hi there everyone,
I am a bit confused these days due to some problem I am having. Its not a
technical problem. Asterisk is working fine. Most of the users are happy,
but some handful of users are getting calls in the middle of the night even
though they have enabled "Anonymous Call Rejection (blocks calls with no
caller id on asterisk server)" and TIMED DO NOT DISTURB which also blocks
calls unconditionally from 11pm to 6 am. Now the seems to have make it
through to the user still. The caller id of the call is "Asterisk Unknown".
about six users are getting this call only at night time. Asterisk server
has no record of this call in log file or cdr. I have also blocked all
incoming calls coming from unknown ip addresses etc.

Still last night there was a call to a customer. Plz help me figure out the
solution for this problem.

Thanks

-- 
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
E: rizwanhas...@gmail.com
W: www.axvoice.com
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[asterisk-users] phoneprov

[Rizwan Hisham]

Hi List,
Can anyone please tell me how to use the phoneprov.conf to provision my
client's atas. I read the file but dont know how to actually use it.

-- 
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Rizwan Qureshi
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Re: [asterisk-users] clustering

[Rizwan Hisham]

Hello Zeeshan,
How about doing the mixture of what I want to do with your strategy. I mean,
what if we have 3 asterisk servers with distributed registrations and also
have heartbeat installed monitoring all the servers? will that work?

On Mon, Oct 18, 2010 at 9:34 PM, Zeeshan Zakaria  wrote:

> How about setting up a high availability cluster using DRBD and Heartbeat?
> There is some good info on it on the Internet. In this type of setup you
> have two exact same servers running in parallel, and only one has the
> required services up. They keep themselves in sync. When the primary one
> goes down, the secondary instantly takes over. Active calls are though
> dropped, but after that everything is back to normal. There are various
> other options regarding which server will stay primary, or how and which
> services will be used on which server.
>
> Another option I am exploring is using the same thing but in Proxmox with
> DRBD. Somebody told me it could be setup so that even the active calls are
> not dropped. I haven't set it up yet, but will try it when get time.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-10-18 10:59 AM, "Danny Nicholas"  wrote:
>
>   --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham
>
>
> Sent: Monday, October 18, 2010 9:43 AM
>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Subject: Re: [asterisk-users] clustering
>
>
>
> Unfortunately we are too late to switch to Kamailio. I mean we have
> developed our pbx with call features and routing on asterisk only. If we
> switch to some other software that means we will have to redo a lot of
> development again. I was thinking of using DUNDi and distributing the
> registrations on different servers.
>
>
>
> I just dont get one point. lets say if i have 2 users registered on
> different asterisk servers and...
>
> 
>
> Sorry for second post, but I have a Polycom 501 registered to 3 servers.  I
> hit the line button and if the server I pick is down, I don’t get a dial
> tone.  Hope this is useful.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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Re: [asterisk-users] clustering

[Rizwan Hisham]

Unfortunately we are too late to switch to Kamailio. I mean we have
developed our pbx with call features and routing on asterisk only. If we
switch to some other software that means we will have to redo a lot of
development again. I was thinking of using DUNDi and distributing the
registrations on different servers.

I just dont get one point. lets say if i have 2 users registered on
different asterisk servers and one of the server fails (dundi doea not get
anything in return from lookup). But I can still get the contact information
for the user who was registered on the failed server from db (realtime peer)
for incoming calls. But what happens when that user tries to make a outgoing
call? How do I redirect the call to the server which is still working?

On Mon, Oct 18, 2010 at 4:39 PM, Gareth Blades
wrote:

> Use camailio or opensips as the registrar server so it accepts the sip
> registrations. You can have copies running on a couple of boxes using
> either a shared databases or a database on each server configured in
> master-master replication mode. Opensips can be configured to use the
> same database table that asterisk uses for authentication. Then you can
> use the load balancer module to send the call to whichever asterisk box
> has the most free lines.
> Normally you try and use opensips for most things such as call routing
> and registrations and leave asterisk to do the application type stuff
> such as conference calls and voicemail.
>
> Rizwan Hisham wrote:
> > anymore ideas anyone please?
> >
> > On Fri, Oct 15, 2010 at 8:36 PM, Josef Grand  > <mailto:josef.gra...@gmail.com>> wrote:
> >
> > use camailio for SIP SLB
> > sip load balancer
> >
> >
> > - Original Message -
> > *From:* Rizwan Hisham <mailto:rizwanhas...@gmail.com>
> > *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> > <mailto:asterisk-users@lists.digium.com>
> > *Sent:* Thursday, October 14, 2010 5:01 PM
> > *Subject:* [asterisk-users] clustering
> >
> > Hi all,
> > I am planning to do clustering for my company's asterisk
> > servers. I dont know much about it, just read some articles on
> > the internet and learned how to use DUNDi and some basic
> > information about clustering.
> > What I need to know is:
> > 1. can i register end user with multiple asterisk servers at a
> time?
> > 2. If not, Can I re-route registeration requests to different
> > servers using 1 asterisk server as a gateway and multiple
> > clustered asterisk servers behind it?
> >
> > cheers
> > Thanks in advance
> >
> > --
> > Best Regards
> > Rizwan Qureshi
> >
> >
> >
> 
> >
> > --
> >
> _
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> >
> >
> > __ Information provenant d'ESET NOD32 Antivirus, version
> > de la base des signatures de virus 5531 (20101014) __
> >
> > Le message a iti virifii par ESET NOD32 Antivirus.
> >
> > http://www.eset.com
> >
> >
> >
> > __ Information provenant d'ESET NOD32 Antivirus, version de
> > la base des signatures de virus 5531 (20101014) __
> >
> > Le message a été vérifié par ESET NOD32 Antivirus.
> >
> > http://www.eset.com
> >
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> > --
> > Best Regards
> > Rizwan Qureshi
> >
> >
>
>
> --
> ___

Re: [asterisk-users] clustering

[Rizwan Hisham]

anymore ideas anyone please?

On Fri, Oct 15, 2010 at 8:36 PM, Josef Grand  wrote:

>  use camailio for SIP SLB
> sip load balancer
>
>
> - Original Message -
> *From:* Rizwan Hisham 
> *To:* Asterisk Users Mailing List - Non-Commercial 
> Discussion
> *Sent:* Thursday, October 14, 2010 5:01 PM
> *Subject:* [asterisk-users] clustering
>
> Hi all,
> I am planning to do clustering for my company's asterisk servers. I dont
> know much about it, just read some articles on the internet and learned how
> to use DUNDi and some basic information about clustering.
> What I need to know is:
> 1. can i register end user with multiple asterisk servers at a time?
> 2. If not, Can I re-route registeration requests to different servers using
> 1 asterisk server as a gateway and multiple clustered asterisk servers
> behind it?
>
> cheers
> Thanks in advance
>
> --
> Best Regards
> Rizwan Qureshi
>
>
>  --
>
> --
> _
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>
>
> __ Information provenant d'ESET NOD32 Antivirus, version de la base
> des signatures de virus 5531 (20101014) __
>
> Le message a iti virifii par ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
> __ Information provenant d'ESET NOD32 Antivirus, version de la base
> des signatures de virus 5531 (20101014) __
>
> Le message a été vérifié par ESET NOD32 Antivirus.
>
> http://www.eset.com
>
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>



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[asterisk-users] clustering

[Rizwan Hisham]

Hi all,
I am planning to do clustering for my company's asterisk servers. I dont
know much about it, just read some articles on the internet and learned how
to use DUNDi and some basic information about clustering.
What I need to know is:
1. can i register end user with multiple asterisk servers at a time?
2. If not, Can I re-route registeration requests to different servers using
1 asterisk server as a gateway and multiple clustered asterisk servers
behind it?

cheers
Thanks in advance

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Re: [asterisk-users] Alert-Info advice

[Rizwan Hisham]

I use the following syntax for sipura i think, and it works fine for me.

exten=> s,1010,SipAddHeader(Alert-Info: )
exten=> s,1020,SipAddHeader(Alert-Info: )
exten=> s,1030,SipAddHeader(Alert-Info: )
exten=> s,1040,SipAddHeader(Alert-Info: )
exten=> s,1050,SipAddHeader(Alert-Info: )

On Wed, Sep 29, 2010 at 12:38 PM, Philipp von Klitzing <
klitz...@pool.informatik.rwth-aachen.de> wrote:

> Hi!
>
> > Just out of interest, have you ever got this working?
>
> Yes, sure.
>
> >  Mine just isn't but I'm starting to think that my mp3 to 8000Hz Mono
> > 16 bit wav files is a bit dodgy
>
> Very well possible. Also look at the individual "identity x"
> configuration and consider to select "Custom ringtone", then enter the
> URL for the wav file in question. That is a good and easy way to test if
> that specifc wav file actually works - also check the http log and/or the
> snom log.
>
> Philipp
>
>
> --
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Re: [asterisk-users] Difference

[Rizwan Hisham]

Thanks for sharing all of your thoughts and information. If anyone knows a
good article about asterisk 1.8 then please let me know about it. I have
read the presentation by Kevin Fleming but more information is always good.

Cheers

On Wed, Oct 6, 2010 at 10:28 AM, Miguel Molina wrote:

>  I find 1.6.2.13 version is stable for trunk call routing, and it should be
> too for basic call center use. The asterisk team has made some architectural
> improvements (moving to astobj2 a lot of internal structures, and much more
> you may not see from a user perspective) but given the several environment
> and different use cases, fear to upgrade or proven 1.4 stability for the
> job, the people usually don't upgrade or make it slowly with a lot of
> previous tests before making the jump.
>
> If you use FAX, I recommend you 1.6.2 or later. The app_fax module is far
> better than the ast-agx-addons for 1.4.
>
> The good old (now unsupported) 1.2 works for many people, ask Steve.
>
> So it's up to you.
>
> Cheers,
>
> --
> Ing. Miguel Molina
> Grupo de Tecnología
> Millenium Phone Center
>
> El 06/10/10 11:04, Zeeshan Zakaria escribió:
>
> For a production environment, 1.4 is the most stable, and it has everything
> one needs to setup a telecom platform. As per my understanding 1.6 never got
> the same recognition for stability as 1.4, plus it doesn't have any
> significant advantages over 1.4. The newer version 1.8 series might be my
> next jump once it'll be out of beta, but at this time it should not be used
> in a production environment. Many of us still use 1.4 in production and if
> you are just starting, this'll be your best choice.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-10-06 11:54 AM, "Danny Nicholas"  wrote:
>
>   From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Be...
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham
> *Sent:* Wednesday, October 06, 2010 10:44 AM
>
>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Difference
>
>
>
>
>
> Is there any major architectural difference between 1.4 and 1.8?
>
> The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from
> , to |) and the AGI structure is enhanced.  If you don’t use AGI’s, a
> qualified “not really”.
>
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Re: [asterisk-users] Difference

[Rizwan Hisham]

Back in the days i heard that they have changed the architecture in 1.6 and
its a lot better than 1.4 (6 times better call handling and robust
architecture, someone told me). If they have decided to take the 1.6
architecture to the next level in the new 1.8 version then its a good thing.

On Wed, Oct 6, 2010 at 9:58 PM, Steve Edwards wrote:

> On Wed, 6 Oct 2010, Rizwan Hisham wrote:
>
> > Is there any major architectural difference between 1.4 and 1.8?
>
> Nope. The developer's just got tired of typing .4
>
> Of course, the joke's on them -- 1.8 is only .4 better than 1.4.
>
> --
> 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] Difference

[Rizwan Hisham]

Is there any major architectural difference between 1.4 and 1.8?

On Wed, Oct 6, 2010 at 7:17 PM, Danny Nicholas  wrote:

>   --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham
> *Sent:* Wednesday, October 06, 2010 7:15 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Difference
>
>
>
> Hi All,
> Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk
> versions.
>
> Thanks
>
> --
> Best Regards
> Rizwan Qureshi
>
>  In a nutshell, 1.4 is the oldest and most stable, 1.6 is the current and
> 1.8 is the beta version of Asterisk.  This is a gross over-simplification,
> but if you “know nothing”, 1.4 is going to give you the fewest headaches and
> if you “have to have the latest” 1.6 or 1.8 is the way to go.  The
> ChangeLogs on Asterisk.org will give you a detailed difference.
>
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[asterisk-users] Difference

[Rizwan Hisham]

Hi All,
Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk
versions.

Thanks

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Re: [asterisk-users] MYSQL ADDON INSTALLATION ERROR

[Rizwan Hisham]

Thank you all. It is now installed.

On Wed, Oct 6, 2010 at 5:04 PM, Steve Howes wrote:

>
> On 6 Oct 2010, at 11:35, Rizwan Hisham wrote:
>
> > Hi All,
> > Please refresh my memory. I am trying to install asterisk after 2 years.
> I hav'nt used it since 2008 (version 1.4.2). Now I am trying to install
> 1.8.0-rc2 on centos 5.5 but getting the following errors.
> > 
> > Plz help.
>
> You need mysql-devel. You might also find that most things are case
> sensitive, maybe your malfunctioning caps-lock is causing problems? ;)
>
> S
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[asterisk-users] MYSQL ADDON INSTALLATION ERROR

[Rizwan Hisham]

Hi All,
Please refresh my memory. I am trying to install asterisk after 2 years. I
hav'nt used it since 2008 (version 1.4.2). Now I am trying to install
1.8.0-rc2 on centos 5.5 but getting the following errors.

app_mysql.c:33:25: error: mysql/mysql.h: No such file or directory
app_mysql.c: In function ‘mysql_ds_destroy’:
app_mysql.c:135: warning: implicit declaration of function ‘mysql_close’
app_mysql.c:138: warning: implicit declaration of function
‘mysql_free_result’
app_mysql.c: In function ‘aMYSQL_connect’:
app_mysql.c:319: error: ‘MYSQL’ undeclared (first use in this function)
app_mysql.c:319: error: (Each undeclared identifier is reported only once
app_mysql.c:319: error: for each function it appears in.)
app_mysql.c:319: error: ‘mysql’ undeclared (first use in this function)

I think i have seen these errors before and did manage to get rid of them
but I cant remember how i did it and even dont remember the reason for these
errors. Looks like a header file for mysql addon is missing which is
actually missing (i have checked). How am I suppose to find it?

Plz help.

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[asterisk-users] MacroExclusive crashed asterisk

[Rizwan Hisham]

Hi all,
I think running the macroexclusive application if it is run after hangup (on
h extension) crashes asterisk. This has happened a lot of times since i
started using the macro exclusive application.

There is a situation in my dialplan when after the user hangsup the call, i
execute the macro exclusive application and in that macro i disconnect the
mysql connection which was made during that call. And asterisk crashes on
that extension most of the time (not all the time though). Are we supposed
to not use the macroexclusive application on dead or hungup channels?


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Re: [asterisk-users] two sip listening ports for single asterisk

[Rizwan Hisham]

Hi guys,
I told my network admin to do what was advised in this thread. It works very
well for incoming calls but outgoing calls hangup exactly after 20 secs
everytime while displaying the following message on cli:

v[Nov 19 10:26:26] WARNING[18617]: chan_sip.c:1910 retrans_pkt: Maximum
retries exceeded on transmission [EMAIL PROTECTED] for seqno
102 (Critical Response)
[Nov 19 10:26:26] WARNING[18617]: chan_sip.c:1927 retrans_pkt: Hanging up
call [EMAIL PROTECTED] - no reply to our critical packet.
  == Spawn extension (macro-rating, s, 104) exited non-zero on
'SIP/saad-a8b83300' in macro 'rating'
  == Spawn extension (macro-rating, s, 104) exited non-zero on
'SIP/saad-a8b83300'

Also, this happens only with certain network conditions. in my office the
outgoing call hangsup everytime but when i dial from home, both incoming and
outgoing calls are fine. So i guess the problem is with user network
configuration. asterisk for other users who are using different port to
register is listening without any problems.


here is my network scenario(from where i make call):
we have a zyxel dsl modem connected to our ISP line
then we have a D-Link switch connected to the dsl modem
then we have sipura 2100 connected to that switch
IP addresses are in the range of 192.168.0.0
NAT type on router = SUA ONLY

On Thu, Nov 20, 2008 at 5:16 PM, Matthew J. Roth <[EMAIL PROTECTED]> wrote:

> Mike wrote:
> > I tried using this iptables sample, and did not see duplicate packets
> > on '--to-ports' port
> >
> > Has some verified this is working for them?
> >
> > I listened on both ports with tcpdump command.
>
> Mike,
>
> I can confirm that it's working.  Admittedly, I never looked at the
> packets with tcpdump because this *just worked* for me.  Calls that were
> sent to both ports (5060 and 5062) made it to Asterisk which was only
> listening on port 5060.  What's your experience with actual calls?
>
> As the original poster, I understand if you want third-party
> verification.  I *thought* this was a slamdunk but I'm not an iptables
> guru so I'd like it, too.
>
> What does the output of "iptables-save" and "lsmod" look like?  Here's
> mine, trimmed for relevancy:
>
> [EMAIL PROTECTED] ~]# iptables-save
> # Generated by iptables-save v1.3.5 on Thu Nov 20 12:03:21 2008
> *nat
> :PREROUTING ACCEPT [5579:1727747]
> :POSTROUTING ACCEPT [1943:176116]
> :OUTPUT ACCEPT [1943:176116]
> -A PREROUTING -i eth2 -p udp -m udp --dport 5062 -j REDIRECT --to-ports
> 5060
> COMMIT
> # Completed on Thu Nov 20 12:03:21 2008
>
> [EMAIL PROTECTED] ~]# lsmod
> Module  Size  Used by
> ip_conntrack_netbios_ns36033  0
> ipt_REDIRECT   35009  1
> xt_tcpudp  36417  1
> iptable_nat40773  1
> ip_nat 53101  2 ipt_REDIRECT,iptable_nat
> ip_conntrack   91237  3
> ip_conntrack_netbios_ns,iptable_nat,ip_nat
> nfnetlink  40457  2 ip_nat,ip_conntrack
> ip_tables  55329  1 iptable_nat
> x_tables   50377  4
> ipt_REDIRECT,xt_tcpudp,iptable_nat,ip_tables
>
> Regards,
>
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
>
>
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[asterisk-users] two sip listening ports for single asterisk

[Rizwan Hisham]

Hi all,
We are planning to shift our sip users from one platform to another.
(basically from one asterisk server to another). the problem we are facing
is that both asterisk servers are using different ports to listen for sip.
and both have live customers on them.  provisioning their ata's is not a
good option for us coz of our settup. we cant just ask the customers to
change their ports for registration (many of them dont know what a port is)
Is it possible to make single asterisk server listen on two different ports?

if there is any other option we can use, plz inform me about it.

Thanx in advance.
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Re: [asterisk-users] Fax with asterisk

[Rizwan Hisham]

The fax is originated from a fax machine connected to an ata which supports
t38.

On Wed, Sep 24, 2008 at 11:54 PM, C F <[EMAIL PROTECTED]> wrote:

> On Wed, Sep 24, 2008 at 5:43 AM, Rizwan Hisham <[EMAIL PROTECTED]>
> wrote:
> > Hi all,
> > Sorry to interrupt. I need some help regarding fax passthru mode.
> >
> > We are trying to configure fax passthru mode in asterisk using sip. For
> out
> > of network calls/fax we use trunk configuration. i am using asterisk
> 1.4.2.
> > The user has to use fax machine connected to their ata and dial the
> callee
> > number, the call is originated just like a regular voice call. have not
> > defined any special context for sending faxes. Have enabled t38 and
> > canreinvite in peer/user and trunk configuration. But the fax is not
> going
> > thru. Our service provider does support fax passthru. Following is the
> trunk
> > and user/peer configuration:
>
> They support passthru, and the originating send fax is what? PSTN? or
> VoIP ATA with t38 support?
> There has to one that does the t38, if the point where it gets
> converted to VoIP does not support t38 then passthru will not help
> you.
>
> >
> > TRUNK CONF
> > [TRUNK-OUT]
> > type=peer
> > host=XXX
> > port=5060
> > context=default
> > country=us
> > dtmfmode=rfc2833
> > restrictcid=no
> > canreinvite=yes
> > insecure=no
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=g729
> > allow=gsm
> > promiscredir=yes
> > t38_udptl=yes
> >
> > USER/PEER
> >
> > [abc]
> > username=abc
> > type=friend
> > secret=123
> > qualify=25000
> > nat=yes
> > mailbox=12129339037
> > insecure=port,invite
> > incominglimit=2
> > outgoinglimit=2
> > intl_trunk=TRUNK-OUT
> > local_trunk=TRUNK-OUT
> > host=dynamic
> > dtmfmode=inband
> > context=uscan
> > canreinvite=yes
> > callerid="Rizwan Qureshi" <122>
> > accountcode=1:0:abc
> > amaflags=default
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=gsm
> > t38_udptl=yes
> >
> >
> > Any solutions?
> >
> > On Tue, Sep 23, 2008 at 10:21 PM, Andrew Joakimsen <[EMAIL PROTECTED]>
> > wrote:
> >>
> >> On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro
> >> <[EMAIL PROTECTED]> wrote:
> >> > ATAs work OK I guess, just make sure to use a loss less codec such as
> >> > ULAW.
> >>
> >> Since the OP stated he is using E1 lines then he should probably be
> >> using alaw instead.
> >>
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> >
> >
> >
> > --
> > Best Regards
> > Rizwan Hisham
> >
> >
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Re: [asterisk-users] wad happen if there is nothing wrong in conf but still can't make calls?

[Rizwan Hisham]

you must share your configuration with us. otherwise we cant even make a
wild guess.

On Mon, Sep 22, 2008 at 7:48 PM, Cindy Tan <[EMAIL PROTECTED]> wrote:

>  may i noe wad can i do because my asterisk is working fine but the calls
> cannot proceed between 2 asterisk servers.
> hope anyone can help me solve this major problem.
>
> thanks a lot in advance
>
> Regards
>
> --
> Make the most of what you can do on your PC and the Web, just the way you
> want. Windows Live <http://www.get.live.com/wl/all>
>
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Re: [asterisk-users] AGI and prepaid billing

[Rizwan Hisham]

We have done it too. www.axvoice.com

On Tue, Sep 23, 2008 at 3:39 PM, Benjamin Jacob <[EMAIL PROTECTED]>wrote:

>
> Hi Bilal,
> Yes it is definitely possible. And I've done it myself for a couple of our
> clients.
> Does that answer your two questions?
>
> cheers
> - Ben.
>
>
>
> --- On Tue, 9/23/08, bilal ghayyad <[EMAIL PROTECTED]> wrote:
>
> > From: bilal ghayyad <[EMAIL PROTECTED]>
> > Subject: [asterisk-users] AGI and prepaid billing
> > To: asterisk-users@lists.digium.com
> > Date: Tuesday, September 23, 2008, 9:52 AM
> > Hi All;
> >
> > Did anyone do an prepaid billing application via AGI? I
> > would like to know if that is possible.
> >
> > Regards
> > Bilal
> >
> >
> >
> >
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>
>
>
>
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Re: [asterisk-users] extension definition

[Rizwan Hisham]

You maybe using wrong username. If the user is defined in sip, you should be
able to register using the correct username and password. Also, see if
asterisk is listening on a defferent sip port instead of default 5060. If
its different use that port.

On Wed, Sep 24, 2008 at 3:32 AM, michel freiha <[EMAIL PROTECTED]> wrote:

> Hello Eric,
> i didwhat you asked me to do but i'm getting "Notfound" sip message when
> trying to register
>
> regrads
>
>
>
> On Tue, Sep 23, 2008 at 9:56 PM, Eric ManxPower Wieling <[EMAIL 
> PROTECTED]>wrote:
>
>> This is done in sip.conf, iax.conf, etc, not in extensions.conf.  By the
>> time a call gets to extensions.conf it must already be authenticated.
>>
>> Assume the username is robertdobbs and the ip is 209.17.71.61
>>
>> In sip.conf you would have something like this:
>>
>> [robertdobbs]
>> deny=0.0.0.0/0
>> permit=209.17.71.61 <http://0.0.0.0/0permit=209.17.71.61>
>> 
>>
>>
>>
>> michel freiha wrote:
>> > Hi all,
>> > I need please the exact extension definition under extensions.conf that
>> > accepts any call coming from an appropriate username and Ip
>> address...This
>> > mean that the authentication should be done on username and IP address
>> >
>> > Regards
>> >
>> >
>> >
>> > 
>> >
>> > ___
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>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
>> --
>> Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
>> T-1, PRI, Frame Relay, Linux, and network design.  Based near
>> Birmingham, AL.  Now accepting clients worldwide.
>>
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>
>
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Re: [asterisk-users] Fax with asterisk

[Rizwan Hisham]

Hi all,
Sorry to interrupt. I need some help regarding fax passthru mode.

We are trying to configure fax passthru mode in asterisk using sip. For out
of network calls/fax we use trunk configuration. i am using asterisk
1.4.2.The user has to use fax machine connected to their ata and dial
the callee
number, the call is originated just like a regular voice call. have not
defined any special context for sending faxes. Have enabled t38 and
canreinvite in peer/user and trunk configuration. But the fax is not going
thru. Our service provider does support fax passthru. Following is the trunk
and user/peer configuration:

TRUNK CONF
[TRUNK-OUT]
type=peer
host=XXX
port=5060
context=default
country=us
dtmfmode=rfc2833
restrictcid=no
canreinvite=yes
insecure=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
promiscredir=yes
t38_udptl=yes

USER/PEER

[abc]
username=abc
type=friend
secret=123
qualify=25000
nat=yes
mailbox=12129339037
insecure=port,invite
incominglimit=2
outgoinglimit=2
intl_trunk=TRUNK-OUT
local_trunk=TRUNK-OUT
host=dynamic
dtmfmode=inband
context=uscan
canreinvite=yes
callerid="Rizwan Qureshi" <122>
accountcode=1:0:abc
amaflags=default
disallow=all
allow=ulaw
allow=alaw
allow=gsm
t38_udptl=yes


Any solutions?

On Tue, Sep 23, 2008 at 10:21 PM, Andrew Joakimsen <[EMAIL PROTECTED]>wrote:

> On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro
> <[EMAIL PROTECTED]> wrote:
> > ATAs work OK I guess, just make sure to use a loss less codec such as
> ULAW.
>
> Since the OP stated he is using E1 lines then he should probably be
> using alaw instead.
>
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Re: [asterisk-users] Asterisk mysql CDR

[Rizwan Hisham]

You can use the ResetCDR() application with the w flag in it after you get
the unavailable, busy or etc message from the callee. It will store the cdr
of that call and after forwarding to mobile, that cdr will be dumped again.

On Wed, Sep 24, 2008 at 8:26 AM, Nhadie <[EMAIL PROTECTED]> wrote:

> hi,
>
> i'm using this macro to dial an extension and forward to a mobile if
> unavailable,busy or noanswer
>
> exten => 100,1,Macro(dial-ext|SIP/${EXTEN}|vm-100|moh-100)
> exten => 100,2,Goto(100-${DIALSTATUS}|1)
> exten => 100-BUSY,1,Macro(dialout-local-mobile|91234567)
> exten => 100-BUSY,2,Voicemail([EMAIL PROTECTED]|u)
> exten => 100-CONGESTION,1,Macro(dialout-local-mobile|91234567)
> exten => 100-CONGESTION,2,Voicemail([EMAIL PROTECTED]|u)
> exten => 100-NOANSWER,1,Macro(dialout-local-mobile|91234567)
> exten => 100-NOANSWER,2,Voicemail([EMAIL PROTECTED]|u)
>
> my prob is on the CDR, from extension 500 i called 100, 100 is not
> online so it should forward it to my mobile
>
> but on the cdr it shows like this:
>
>  FromTo
> 500 100-CHANUNAVAIL
>
> should it be like
>
>  FromTo
> 500 91234567
>
> or
>
>  FromTo
> 100 91234567
>
> any idea how to fix those?
>
> regards,
> nhadie
>
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[asterisk-users] rxfax and txfax

[Rizwan Hisham]

Hi all,
I want to configure my asterisk for sending and receiving faxes. I see in my
sip.conf that i have to enable the t.38 capability. I have done that but the
rxfax and txfax applications are not installed. They are not listed in
applications when i do make menuselect. i have searched in voip-info wiki,
found a 
page<http://www.voip-info.org/wiki/index.php?page_id=2583&tk=99b8d086f0f28f4c1542&comments_page=1>but
the links given on that page for downloading the applications are not
working. I am using asterisk 1.4.2, i thaught the missing applications maybe
included in latest version of asterisk but they are not, already downloaded
and checked in asterisk 1.4.21.

How can i install these applications. Are there anyother components required
to make my asterisk a fax-passthru system.

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Re: [asterisk-users] SIP URI Forwarding

[Rizwan Hisham]

thats what i am passing

exten=> 456,1,Dial(SIP/adf:[EMAIL PROTECTED]:9060

adf is username and 123 is the password



On Wed, Sep 17, 2008 at 6:01 PM, Alex Balashov <[EMAIL PROTECTED]>wrote:

>
> If there is a secret= on the receiving peer, the sending peer needs to
> provide that secret.  Along with a username.
>
> Rizwan Hisham wrote:
>
> > Hi all,
> > I am having a problem with sip uri incoming calls. I have 2 asterisk
> > servers both are 1.4.2. <http://1.4.2.> i dial sip uri from one asterisk
> > server which sends the call to the other asterisk server by seeing its
> > domain name in the uri. Invite reaches the recieving asterist server but
> > the call is not autenticated. Everytime i see the following NOTICE on
> > the asterisk server (caller end)
> >
> > [Sep 17 15:38:24] NOTICE[4594]: chan_sip.c:11968 handle_response_invite:
> > Failed to authenticate on INVITE to '"rizwan"  > <http://sip:[EMAIL PROTECTED]:9860>>;tag=as089d4adb'
> >
> > My dialplan on caller end is:
> >
> > [directcall]
> > exten=> 123,1,Dial(SIP/abc:[EMAIL PROTECTED]:9060
> > <http://abc:[EMAIL PROTECTED]:9060>)
> > exten=> 123,2,Hangup()
> >
> > exten=> 456,1,Dial(SIP/adf:[EMAIL PROTECTED]:9060
> > <http://adf:[EMAIL PROTECTED]:9060>)
> > exten=> 456,2,Hangup()
> >
> > SIP general settings on receiving end are:
> >
> > [general]
> > context=uricall-incoming
> > allowoverlap=no
> > bindport=9060
> > bindaddr=0.0.0.0 <http://0.0.0.0>
> > srvlookup=yes
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=g729
> > allow=gsm
> > relaxdtmf=yes
> > useragent=Asterisk PBX
> > dtmfmode = rfc2833
> > nat=no
> > canreinvite=yes
> >
> > peer settings on receiving end:
> >
> > [adf]
> > username=adf
> > type=friend
> > secret=XXX
> > qualify=25000
> > nat=yes
> > insecure=port,invite
> > host=dynamic
> > dtmfmode=rfc2833
> > context=sipuri-incoming
> > canreinvite=yes
> > callerid="adf xyz" <123>
> > accountcode=6:0:adf
> > amaflags=default
> > disallow=all
> > allow=g729
> > allow=ulaw
> > allow=alaw
> > allow=gsm
> >
> > am i doing something wrong here?
> >
> >
> > --
> > Best Regards
> > Rizwan Hisham
> >
> >
> > 
> >
> > ___
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>
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
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Re: [asterisk-users] dtmf passthru

[Rizwan Hisham]

Alex,
So u mean DTMF is by default passed thru already.

Benjamin,
Well i dont want to enforce my asterisk dtmf setting on any call. What i
want is whatever the user has set the dtmf mode in his ata or softphone,
that should be used to pass the dtmf signals on to the callee.

On Wed, Sep 17, 2008 at 6:05 PM, Alex Balashov <[EMAIL PROTECTED]>wrote:

> What is "DTMF passthru?"
>
> DTMF is regenerated by default.  If the DTMF mode is inband, it's simply
> part of the audio stream.  If it uses named RTP events, those are
> regenerated on the other call leg.
>
> Rizwan Hisham wrote:
>
> > hi all,
> > Is there an option of dtmf passthru mode in asterisk. If yes, how can i
> > do it?
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
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[asterisk-users] dtmf passthru

[Rizwan Hisham]

hi all,
Is there an option of dtmf passthru mode in asterisk. If yes, how can i do
it?


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[asterisk-users] SIP URI Forwarding

[Rizwan Hisham]

Hi all,
I am having a problem with sip uri incoming calls. I have 2 asterisk servers
both are 1.4.2. i dial sip uri from one asterisk server which sends the call
to the other asterisk server by seeing its domain name in the uri. Invite
reaches the recieving asterist server but the call is not autenticated.
Everytime i see the following NOTICE on the asterisk server (caller end)

[Sep 17 15:38:24] NOTICE[4594]: chan_sip.c:11968 handle_response_invite:
Failed to authenticate on INVITE to '"rizwan" ;tag=as089d4adb'

My dialplan on caller end is:

[directcall]
exten=> 123,1,Dial(SIP/abc:[EMAIL PROTECTED]:9060)
exten=> 123,2,Hangup()

exten=> 456,1,Dial(SIP/adf:[EMAIL PROTECTED]:9060)
exten=> 456,2,Hangup()

SIP general settings on receiving end are:

[general]
context=uricall-incoming
allowoverlap=no
bindport=9060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
relaxdtmf=yes
useragent=Asterisk PBX
dtmfmode = rfc2833
nat=no
canreinvite=yes

peer settings on receiving end:

[adf]
username=adf
type=friend
secret=XXX
qualify=25000
nat=yes
insecure=port,invite
host=dynamic
dtmfmode=rfc2833
context=sipuri-incoming
canreinvite=yes
callerid="adf xyz" <123>
accountcode=6:0:adf
amaflags=default
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm

am i doing something wrong here?


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[asterisk-users] strange transfer problem

[Rizwan Hisham]

Hi all,
I am having a strange problem with my asterisk server. When i dial an
outside tollfree number, if there is a menu for example press 1 for support,
press 2 for sales etc, after pressing any given option as the system begins
to transfer me the call hangs up. I have tried it so many times on different
tollfree numbers but the problem remains only when i dial from my asterisk
box. The call is disconnected with a normal call clearing (hangup cause 16).
But when i dial from my cell phone or any other line, the call is transfered
without any problem.

I also checked sip debug and core debug for different calls. I think it has
something to do with strict routing,  the only strange message i get on the
cli is:

[Sep  4 09:10:39] DEBUG[14929]: chan_sip.c:5690 reqprep: Strict routing
enforced for session [EMAIL PROTECTED]

This message does not appear for other calls (when there is no transfering)

I googled a little on strict and loose routing but i did not get it. maybe
someone here can help me solve this problem.

VERSIONS
asterisk 1.4.2
zaptel and libpri 1.4.0

I can send you core debug if you want it.

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Re: [asterisk-users] callfiles/manager api originate call fails

[Rizwan Hisham]

Actually both calls have to be originated to the outside world. Thats why im
using @TRUNK-OUT, when the first call is answered only then the call goes to
a context. Thats where the problem is, the first call does not originate so
i cant throw it to any context.

On Thu, Aug 21, 2008 at 8:47 PM, Anthony Francis <[EMAIL PROTECTED]>wrote:

>
>
> Rizwan Hisham wrote:
> > Hi all,
> > asterisk is giving me tough time. its been 3 days I am trying to
> > originate outgoing call using manager api/callfiles.
> >
> I would say remove the "@TRUNK-OUT" part and make sure that the context
> you send the call to knows about sending calls to the outside world.
>
> --
> Thank you and have any kind of day you want,
>
> Anthony Francis
>
>
>
>
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[asterisk-users] callfiles/manager api originate call fails

[Rizwan Hisham]

Hi all,
asterisk is giving me tough time. its been 3 days I am trying to originate
outgoing call using manager api/callfiles. both seem to work fine when i
originate a call for a local peer, but if i try originating a call outside
using a trunk thats when everything goes wrong. It does originate the call
but the call does not go through to the desired endpoint. The trunk
configuration is correct as all the other calls from users are fine. Am here
for any suggestion. How can i make it work. If anyone knows anyother
technique to originate auto calls from asterisk i'll be happy to try them
out.

 I am using the following manager command,

fputs($socket, "Action: Originate\r\n");

//fputs($socket, "Channel: SIP/abc\r\n");

fputs($socket, "Channel: SIP/".$txt_your_number."@TRUNK-OUT\r\n");

fputs($socket, "Context: webcall\r\n");

fputs($socket, "Exten:
932\r\n");
fputs($socket, "Priority: 1\r\n");

fputs($socket, "CallerID:
WebCall<932>\r\n");
fputs($socket, "Timeout: 3\r\n");

fputs($socket, "Variable: ID=" . $id . "|accountcode=7:0:webcall|sec=" .
$min . "|dialnum=" . $txt_to_number . "|source_num=" . $txt_your_number .
"|calldate=" . date("Y-m-d H:i:s") . "\r\n\r\n");


and my callfile contents are:

Channel: SIP/TRUNK-OUT/$DIALNUM
CallerID: Webcall<932>
MaxRetries: 2
RetryTime: 10
WaitTime: 35
Account: 7:0:webcall
Context: webcall
Extension: 932
Priority: 1
Set: ID=".$id."
Set: accountcode=7:0:webcall
Set: sec=".$allowed_secs."
Set: dialnum=".$dialnum."\
et: source_num=".$srcnum."
Set: calldate=".$calldate." ".$calltime."\n";

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Re: [asterisk-users] shared mysql connection in dialplan

[Rizwan Hisham]

have done it, and its working fine. but still expecting to receive some new
ideas.

On Wed, Aug 6, 2008 at 2:12 PM, Rizwan Hisham <[EMAIL PROTECTED]>wrote:

> hi all,
> i just finished developing some incoming call features in a macro. that
> macro gets executed everytime an incoming call is received and a new mysql
> connection is made using the MYSQL cmd in dialplan. i want to use a single
> mysql connection for every incoming call.
>
> my idea of doing it is like this, i want to get a mysql connection in a
> global variable, just to share the connection with different incoming calls.
> Im not sure if this can be done. I am going to try doing it somehow,
> meanwhile i want your suggestions about how i can share a mysql connection
> with different calls in a dialplan.
>
> I am using asterisk1.4.2 and asterisk addon1.4.0 package for mysql
> connectivity.
>
> Thanx in advance
>
> --
> Best Regards
> Rizwan Hisham
>
>


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[asterisk-users] shared mysql connection in dialplan

[Rizwan Hisham]

hi all,
i just finished developing some incoming call features in a macro. that
macro gets executed everytime an incoming call is received and a new mysql
connection is made using the MYSQL cmd in dialplan. i want to use a single
mysql connection for every incoming call.

my idea of doing it is like this, i want to get a mysql connection in a
global variable, just to share the connection with different incoming calls.
Im not sure if this can be done. I am going to try doing it somehow,
meanwhile i want your suggestions about how i can share a mysql connection
with different calls in a dialplan.

I am using asterisk1.4.2 and asterisk addon1.4.0 package for mysql
connectivity.

Thanx in advance

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Re: [asterisk-users] Need help with implementing prepaid in asterisk

[Rizwan Hisham]

You can calculate users remaining minutes according to his remaining balance
and then set the Absolute timeout for his every outgoing call using
the Timeout(absolute)
= X 
<http://www.voip-info.org/wiki/index.php?page=Asterisk+func+timeout>variable.
This will hangup the call after X seconds
And if there users balance is in negative or zero then just bypass the dial
statement in your dialplan and land on Hangup or Playback before hangup to
let the user know why his call is not being connected.

All of these changes you have to make in extensions.conf

On Tue, Jul 29, 2008 at 11:59 AM, Ian Coetzee <[EMAIL PROTECTED]>wrote:

> Hi all
>
> I am trying to implement a prepaid dialing system on our asterisk box. I
> however have a few questions I need to ask. I have written a simple script
> in php to do all the billing.
>
>1. What do I need to user to cut off the users in mid call
>2. What do I need to insert into my dialplan to deny a user to call
>
> if you need any config files I will send them, seeing as I dont know what
> files to send.
>
> If you can point me to a howto I will be more gratefull.
>
> Regards
> Ian
>
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Re: [asterisk-users] Help With dial plan

[Rizwan Hisham]

maybe the user is dialing something other than 3000 and that extension is
not registered on your asterisk. just a wild guess.

On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku <[EMAIL PROTECTED]> wrote:

> Hi list,
>
> Have installed trixbox and I am working with a fxo gateway to get fxo calls
> to trixbox. I am using sip to send the calls from the gateway to trixbox. I
> have an extension 3000 on trixbox
>
> on [from-sip-external] on extensions.conf ,I have put the dial plan below.
>
> exten => 3000,1,dial(sip/3000)
> exten=> 3000,2,answer()
> exten => 3000,3,congestion()
> exten=> 3000,4,hangup()
>
>
> this works fine. But I when I put it in the form
>
> exten => _3XXX,1,dial(sip/${EXTEN})
> exten=> _3XXX,2,answer()
> exten =>_3XXX,3,congestion()
> exten=> _3XXX,4,hangup()
>
> the call goes into congestion and I get a busy tone. What could I be doing
> wrong?
>
> James
>
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Re: [asterisk-users] Overlap dialing via SIP

[Rizwan Hisham]

i can see from your dialplan that all the extensions except international
extension are of 12 digits. International extensions are of 13 or more
digits. here is what you can do with the international extensions, all other
extensions remain the same:

[084x]
exten => _9084,1,Macro(dialout-pstn)

[outbound-national]
exten => _90[1-2]X,1,Macro(dialout-pstn)

[087x]
exten => _9087,1,Macro(dialout-pstn)

[0906]
exten => _90906XXX,1,Macro(dialout-pstn)

[outbound-international]
exten => _900XX.,1,Dial(SIP/[EMAIL PROTECTED])
exten => _900XX.,2,Congestion

If you see closely i have put a dot at the end of each international
extension, this will allow you to dial atleast 13 digits. so no need to
crate extension of every length.


On Mon, Jul 21, 2008 at 9:10 PM, Ben Thompson <[EMAIL PROTECTED]> wrote:

> Hi
>
> I have set up an asterisk system which allows the use of Overlap Dialing
> from
> SIP handsets. In order to do this I had to list the various patterns of
> numbers
> which can be dialed in the UK. We also dial with a prefix of '9' for and
> outside
> line so much of my dialplan looks like this :-
>
> [084x]
> exten => _9084,1,Macro(dialout-pstn)
>
> [outbound-national]
> exten => _90[1-2]X,1,Macro(dialout-pstn)
>
> [087x]
> exten => _9087,1,Macro(dialout-pstn)
>
> [0906]
> exten => _90906XXX,1,Macro(dialout-pstn)
>
> ...
>
>
> I was able to download the mappings for 0800 numbers and other special
> ranges
> from the ofcom website and I have incorporated these. For international
> dialing
> I have not been able to find an easy way of doing this so I created the
> folling
> contexts whcih make use of the WaitExten feature :-
>
> [outbound-international]
> exten => _900XX,1,Set(oldexten=${EXTEN})
> exten => _900XX,2,Goto(international-number-length-check,s,1)
>
> [international-number-length-check]
> exten => s,1,Answer
> exten => s,2,WaitExten(8)
>
> exten => _X,1,Set(enddigits=${EXTEN})
> exten => _X,2,NoOp(${TIMESTAMP} ok 13 digits - we dial
> ${oldexten}${enddigits})
> exten => _X,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
> exten => _X,4,Congestion()
> exten => _X,104,Busy()
>
> exten => _XX,1,Set(enddigits=${EXTEN})
> exten => _XX,2,NoOp(${TIMESTAMP} ok 14 digits - we dial
> ${oldexten}${enddigits})
> exten => _XX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
> exten => _XX,4,Congestion()
> exten => _XX,104,Busy()
>
> exten => _XXX,1,Set(enddigits=${EXTEN})
> exten => _XXX,2,NoOp(${TIMESTAMP} ok 15 digits - we dial
> ${oldexten}${enddigits})
> exten => _XXX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
> exten => _XXX,4,Congestion()
> exten => _XXX,104,Busy()
>
> exten => t,1,Dial(${OUTBOUNDTRUNK}/${oldexten})
> exten => t,2,Congestion()
> exten => t,102,Busy()
>
>
> This works fairly well but I have noticed that occasionally the WaitExten
> feature does
> not seem to catch the first digits if they are dialed too quickly. It is
> almost as if
> there is a some sort of delay and the thirteenth digit is sometimes missed.
>
> Can anyone suggest why WaitExten might be ocasionally missing a digit or
> can anyone think
> of a better way of doing this?
>
> Thanks
>
> Ben Thompson
>
>
>
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Re: [asterisk-users] Dial command and its g option

[Rizwan Hisham]

 just add as many extensions as you want under the Dial command extension
keeping the extension number same:

exten => s,n,Dial(SIP/100,100,Ttg)
exten => s,n,Application here



On Thu, Jun 12, 2008 at 3:25 PM, voip crazy <[EMAIL PROTECTED]> wrote:

> I need to execute an action after a call is hangup. I just see the
> command Dial has an option for that, the "g" option.
> I configure the dial command as
>
> exten => s,n,Dial(SIP/100,100,Ttg)
>
> How should I add the line which the command will be executed after the
> dial command in this example?
>
> I don`t how its works, someone could put a example about the way to use it.
>
> Thanks you in advance.
>
> VoipCrazy
>
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[asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)

[Rizwan Hisham]

Hi all,
I have setup an asterisk system which:

   1. recieves incoming sip calls
   2. ask the caller the number they want to dial, and then dial that number
   3. after the caller is done talking and callee hangsup or even if the
   callee does not answer the phone, the caller is asked for another number to
   dial.
   4. And so onuntill the caller hangsup

Everthing above is working fine. But i dont know how to manipulate the cdr
so that every outgoing call for he caller should be logged. I have looked
into ForkCDR but it seems like it can only be used for transfers.

Any ideas how i can solve my multiple cdr problem?
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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

[Rizwan Hisham]

Brent, hope your problems go away soon.

I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We are
using asterisk 1.4.2 for a SIP only based configuration. Currently we have
about 200 SIP users which can cause approximately upto 3 simultaneous calls.
We are mainly concerned about the performance and stability of asterisk when
the load increases. Our server can handle about 100 simultaneous calls
having 3Ghz Dual Intl-Xeon Processor with 2GB of ram using G711 codec, and
around 30 simultanoeus calls using G729 codec. This we are expecting from
the hardware. We are planning to accomodate  about 5,000 users on this
server.

Before the release of 1.6 i heard that its architecture is going to be
different from 1.2 and 1.4. Recently i read an article about
freeswitch<http://freeswitch.org/node/117>,
which explains how its functionality is like asterisk but it can perform
better than asterisk due to its architectural differences. The main
developer for freeswitch is anthony who also codes for asterisk. He
explaines why the architecture of asterisk needs to be changed which
requires massive recoding, but nobody took the step to do it. Thats why he
started freeswitch on his own to redifine the architecture, so that the
performance and reliability of the switch should be better than asterisk. In
his article he has already said that freeswitch beats asterisk by a factor
of 10.

If asterisk architecture is being rewritten in 1.6 to achive the same goal,
then we will be happy to use 1.6 instead of shifting the whole system to
freeswitch. We dont have any problem or issues with 1.4.2 yet. We are mainly
concerned about the its performance when the load increases. If 1.6 is more
reliable under heavy loads then we would like to use it.

If anyone can put some light on this topic, all i can say is thanx for
sharing your thaughts and experiences.



On Thu, Jun 5, 2008 at 1:13 AM, Brent Davidson <[EMAIL PROTECTED]>
wrote:

> Just an update.  I tried updating to the newest Rhino Release firmware
> 1.15 and newest stable driver version 2.2.6.  It works OK with
> zaptel-1.4.9.2 and compiles OK with 1.4.10.1 but when compiled against
> zaptel 1.4.10.1 Asterisk does not see any zap channels.  I'm currently
> running one branch office with the upgraded firmware, driver,
> zaptel-1.4.9.2 and Asterisk-1.4.20.1.  I'll see how everything goes
> there and may upgrade the other offices if it works OK.
>
> Thanks,
> Brent
>
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Re: [asterisk-users] where did the switch statement come from?

[Rizwan Hisham]

where can i find details about both switch statement and dundi.

On Mon, May 19, 2008 at 4:37 PM, Alexander Lopez <[EMAIL PROTECTED]>
wrote:

>  The switch statement allows you to 'include' a context from another
> machine into your machine.
>
>
>
> Problems with it was if the other machine was unavailable, or even slow to
> respond, your machine would hang until it timed out.
>
>
>
> DUNDI has since replaced the functionality of the switch statement and
> given you so much more in return
>
>
>   --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Rizwan Hisham
> *Sent:* Monday, May 19, 2008 6:59 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] where did the switch statement come from?
>
>
>
> Hi all,
> I have looked up the applications and function in asterisk but i could not
> find the help for the switch statement which is used in several places in
> sample extensions.conf file. i am using asterisk 1.4.2. On voip-info.orgthe 
> switch statement seems to be used to connect 2 asterisk servers, but i
> could not find a satisfactory explanation for the this statement. Can
> anybody help me understand the switch statement?
>
> --
> Best Regards
> Rizwan Hisham
>
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[asterisk-users] where did the switch statement come from?

[Rizwan Hisham]

Hi all,
I have looked up the applications and function in asterisk but i could not
find the help for the switch statement which is used in several places in
sample extensions.conf file. i am using asterisk 1.4.2. On voip-info.org the
switch statement seems to be used to connect 2 asterisk servers, but i could
not find a satisfactory explanation for the this statement. Can anybody help
me understand the switch statement?

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Re: [asterisk-users] Problems passing variables from a macro

[Rizwan Hisham]

I haven't used ael but in extension.conf whenever we set a channel variable
we use a SET command just like you used it to set the variable MACRO_RESULT.
try using the set command, if still it does not work then try to initialize
the wrongpin=0 before the dial command, or outside the macro.

On Fri, May 16, 2008 at 12:36 PM, Tobias Ahlander <[EMAIL PROTECTED]>
wrote:

> Good day,
>
> I'm using a dial string as follows:
> Dial(SIP/${phonenumber},30,grM(screen^${pin})L(${30}[:6]));
> When I set a variable in the macro screen, it doesn't get passed back to
> the extension from where the dial was called. I can always put the result in
> the MySQL database, but that feels a bit overkill... the macro looks as
> follows:
>
> macro screen (arg1) {
>
>   Wait(0.2);
>   Read(acceptcall|sounds/pin|7);
>   if(${acceptcall} = ${arg1}) {
> NoOp(connect them);
> wrongpin=0;
>   } else {
> Set(MACRO_RESULT=CONTINUE);
> wrongpin=1;
>   }
>   NoOp(MACRO_RESULT = ${MACRO_RESULT});
>
> }
>
> This is the output from the CLI, and I can see that the wrongpin is set to
> 1, but when I do a NoOp right after leaving the macro, it says its empty...
>
> -- Executing [EMAIL PROTECTED]:36] Dial("SIP/1003-b7619b78",
> "SIP/1203|30|grM(screen^1234)L(30[:6])") in new stack
> -- Limit Data for this call:
>> timelimit  = 30
>> play_warning   = 6
>> play_to_caller = yes
>> play_to_callee = yes
>> warning_freq   = 0
>> start_sound= (null)
>> warning_sound  = beep
>> end_sound  = beep
> -- Called 1203
> -- SIP/1203-08d62408 is ringing
> -- SIP/1203-08d62408 answered SIP/1003-b7619b78
> -- Executing [EMAIL PROTECTED]:1] Set("SIP/1203-08d62408", "arg1=1234")
> in new stack
> -- Executing [EMAIL PROTECTED]:2] Wait("SIP/1203-08d62408", "0.2") in
> new stack
> -- Executing [EMAIL PROTECTED]:3] Read("SIP/1203-08d62408",
> "acceptcall|sounds/pin|7") in new stack
> -- Accepting a maximum of 7 digits.
> --  Playing 'sounds/pin' (language 'en')
> -- User entered '1'
> -- Executing [EMAIL PROTECTED]:4] GotoIf("SIP/1203-08d62408", "0?5:8")
> in new stack
> -- Goto (macro-screen,s,8)
> -- Executing [EMAIL PROTECTED]:8] Set("SIP/1203-08d62408",
> "MACRO_RESULT=CONTINUE") in new stack
> -- Executing [EMAIL PROTECTED]:9] Set("SIP/1203-08d62408", "wrongpin=1")
> in new stack
> -- Executing [EMAIL PROTECTED]:10] NoOp("SIP/1203-08d62408", "Finish
> if-screen-32753") in new stack
> -- Executing [EMAIL PROTECTED]:11] NoOp("SIP/1203-08d62408",
> "MACRO_RESULT = CONTINUE") in new stack
> -- Executing [EMAIL PROTECTED]:37] NoOp("SIP/1003-b7619b78",
> "DIALSTATUS:ANSWER") in new stack
> -- Executing [EMAIL PROTECTED]:38] NoOp("SIP/1003-b7619b78", "wrongpin=") 
> in
> new stack
>
> Is there a good way to pass this variable back to the context "connect"?
>
> Thanks,
> Best regards,
> Tobias
>
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Re: [asterisk-users] Manager API - Setvar not working

[Rizwan Hisham]

Can anybody help in parsing the manager events efficiently? Any ideas?

On Fri, May 9, 2008 at 5:07 PM, Gunārs Grundāns <
[EMAIL PROTECTED]> wrote:

>
>
> On 5/8/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>>
>> On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
>> > Hi all,
>> > I am using a simple perl script to connect with ast manager api. the
>> script
>> > tries to set a channel variable. It extracts the channel name from the
>> > events it recieves after dial command. When i try to set the channel
>> > variable, asterisk responses with an error saying that the channel does
>> not
>> > exist. Can anybody tell me why its doing so, coz i can see on cli that
>> the
>> > channel exists. If i try to set the variable without stating the channel
>> > then it sets the global variable, but i want to set the channel
>> variable.
>> > Anybody has a solution to this problem?
>>
>>
>> In 1.6 you can set a channel variable from a diferent context. I don't
>> think you can do that in 1.4 .
>>
>> --
>>Tzafrir Cohen
>> icq#16849755  jabber:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
>> +972-50-7952406   mailto:[EMAIL PROTECTED]
>> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>>
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>>
>
>
> My guess that Setvar was deprecated in Asterisk 1.4. Use Set instead if you
> are trying to use it in 1.6
> --
> Gunars Grundans
>
> http://freight.lv
>
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Re: [asterisk-users] Manager API - Setvar not working

[Rizwan Hisham]

Thanx a lot.that was it. will never forget to remove the new
character again. Now its working fine.

On Fri, May 9, 2008 at 4:31 PM, Tony Mountifield <[EMAIL PROTECTED]>
wrote:

> In article <[EMAIL PROTECTED]>,
> Rizwan Hisham <[EMAIL PROTECTED]> wrote:
> >
> > same is the case in 1.6, i cant set the variable still.
>
> My guess would be that you have a problem with line endings.
>
> All lines received from the manager interface are terminated with \r\n,
> not just \n. If you only strip the \n off, the channel name you received
> will contain \r at the end. If you then send that back in the setvar
> command with a further \r\n to terminate the line, then Asterisk will
> probably not find a matching channel.
>
> Make sure you strip both \r and \n from incoming lines.
>
> Cheers
> Tony
>
> > On Thu, May 8, 2008 at 8:43 PM, Tzafrir Cohen <[EMAIL PROTECTED]>
> > wrote:
> >
> > > On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
> > > > Hi all,
> > > > I am using a simple perl script to connect with ast manager api. the
> > > script
> > > > tries to set a channel variable. It extracts the channel name from
> the
> > > > events it recieves after dial command. When i try to set the channel
> > > > variable, asterisk responses with an error saying that the channel
> does
> > > not
> > > > exist. Can anybody tell me why its doing so, coz i can see on cli
> that
> > > the
> > > > channel exists. If i try to set the variable without stating the
> channel
> > > > then it sets the global variable, but i want to set the channel
> variable.
> > > > Anybody has a solution to this problem?
> > >
> > > In 1.6 you can set a channel variable from a diferent context. I don't
> > > think you can do that in 1.4 .
> > >
> > > --
> > >   Tzafrir Cohen
> > > icq#16849755  jabber:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> <[EMAIL PROTECTED] <[EMAIL PROTECTED]>>
> > > +972-50-7952406   mailto:[EMAIL PROTECTED]
> > > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
> > >
> > > ___
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> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> >
> > --
> > Best Regards
> > Rizwan Hisham
> >
> > -=-=-=-=-=-
> > [Alternative: text/html]
> > -=-=-=-=-=-
> > -=-=-=-=-=-
> >
> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > -=-=-=-=-=-
>
>
> --
> Tony Mountifield
> Work: [EMAIL PROTECTED] - http://www.softins.co.uk
> Play: [EMAIL PROTECTED] - http://tony.mountifield.org
>
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Re: [asterisk-users] Manager API - Setvar not working

[Rizwan Hisham]

same is the case in 1.6, i cant set the variable still.

On Thu, May 8, 2008 at 8:43 PM, Tzafrir Cohen <[EMAIL PROTECTED]>
wrote:

> On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
> > Hi all,
> > I am using a simple perl script to connect with ast manager api. the
> script
> > tries to set a channel variable. It extracts the channel name from the
> > events it recieves after dial command. When i try to set the channel
> > variable, asterisk responses with an error saying that the channel does
> not
> > exist. Can anybody tell me why its doing so, coz i can see on cli that
> the
> > channel exists. If i try to set the variable without stating the channel
> > then it sets the global variable, but i want to set the channel variable.
> > Anybody has a solution to this problem?
>
> In 1.6 you can set a channel variable from a diferent context. I don't
> think you can do that in 1.4 .
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] AGI asterisk high balance

[Rizwan Hisham]

Well database really is a bottleneck for me. I am currently trying to do
rating stuff in agi using perl. What im doing is i lookup the rate of every
dialed code for every call from the mysql database using the longest match
technique. The longest match technique costs atleast 2-3 mysql queries for
every call untill the dialed code is matched out of 14000 dialcodes. I dont
know how to calculate the exact delay due to execution of agi, but on the
asterisk cli whenever that agi executes, there is a visual delay of about
half a sec to move from the agi extension to the next extension (can anybody
tell me how to calculate the delay).

Now im planning to use the manager api for constant connectivity to mysql
and to enhance the longest match technique. Can anybody help me with this?
Is it a good idea to ue manager api for  lookingup the rate of the live
call?

On Sun, May 4, 2008 at 1:34 PM, Grey Man <[EMAIL PROTECTED]> wrote:

> If you've got anything but trivial AGI loads you should switch to
> FastAGI and put your business logic on a separate server to your
> Asterisk server. I use a deployment where a call could make up to 3
> AGI requests per call before being put through (for things such as
> looking up accountcode, checking account credit, setting PSTN
> callerid). We monitor the time thw whole process takes and on average
> it's less than 100ms on an Asteisk server that peaks at 200
> simultaneous calls (400 bridged) and 3 to 5 call set ups per second.
> The business logic processing the FastAGI   calls is C# and .net which
> means Java would be able to handle it easily as well. The most likely
> bottleneck under high load will be your database.
>
> Regards,
>
> Greyman.
>
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[asterisk-users] Manager API - Setvar not working

[Rizwan Hisham]

Hi all,
I am using a simple perl script to connect with ast manager api. the script
tries to set a channel variable. It extracts the channel name from the
events it recieves after dial command. When i try to set the channel
variable, asterisk responses with an error saying that the channel does not
exist. Can anybody tell me why its doing so, coz i can see on cli that the
channel exists. If i try to set the variable without stating the channel
then it sets the global variable, but i want to set the channel variable.
Anybody has a solution to this problem?

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[asterisk-users] buying cards from pakistan

[Rizwan Hisham]

Hi all,
i want to buy a pci or whatever card for asterisk to plug in my telephone
line into it and use asterisk as a pbx. i have only one telephone line at
home. can you recommend me a simple cheap card which i can buy in pakistan.

I live in pakistan, and i dont know any dealers here who sell asterisk
cards. if someone knows where to buy cards in pakistan, plz tell me about
it.

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Re: [asterisk-users] Asterisk Warning 2512

[Rizwan Hisham]

I just saw the sip debug and its showing that for every notify request,
asterisk is sending a bad request response.

here is the debug

<--- SIP read from 70.80.000.00:1031 --->
NOTIFY sip:69.90.111.11:9060 SIP/2.0
Via: SIP/2.0/UDP 70.80.000.00:1032;branch=z9hG4bK-ade549bd
From: Blake 
>;tag=90115683e082af23o0
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 7741 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA2102-5.2.3
Content-Length: 0

<->
--- (10 headers 0 lines) ---
<--- Transmitting (no NAT) to 70.80.000.00:1031 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 70.80.000.00:1032;branch=z9hG4bK-ade549bd;received=
70.80.000.00
From: Blake 
>;tag=90115683e082af23o0
To: ;tag=as3ef6a439
Call-ID: [EMAIL PROTECTED]
CSeq: 7741 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

Why is it doing so?

On Thu, Apr 17, 2008 at 2:36 AM, Tony Mountifield <[EMAIL PROTECTED]>
wrote:

> In article <[EMAIL PROTECTED]>,
>  Rizwan Hisham <[EMAIL PROTECTED]> wrote:
> > I have been seeing a lot of the following warning messages on my
> asterisk
> > cli. Can naybody tell why these messages are showing up. I am using only
> SIP
> > to make calls from m asterisk.
> >
> > [Apr 17 04:52:24] WARNING[2512]: chan_sip.c:6480
> determine_firstline_parts:
> > Bad request protocol Bad event
> >
> > Also it will be great if anybody can tell where i can find the
> explanation
> > of all the warnig codes and error codes of asterisk if there is any.
>
> The [2512] is not a warning code. It is just the process ID of the
> Asterisk
> process or thread that generated the warning.
>
> The next part of your message (chan_sip.c:6480) shows the source file and
> line number where the error was generated. You can go to that point in
> the file to see what kind of checks it was making. You can also turn on
> SIP debugging at the Asterisk CLI> prompt to see the packets sent to/from
> Asterisk.
>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: [EMAIL PROTECTED] - http://www.softins.co.uk
> Play: [EMAIL PROTECTED] - http://tony.mountifield.org
>
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[asterisk-users] Asterisk Warning 2512

[Rizwan Hisham]

Hi all,
I have been seeing a lot of the following warning messages on my asterisk
cli. Can naybody tell why these messages are showing up. I am using only SIP
to make calls from m asterisk.

[Apr 17 04:52:24] WARNING[2512]: chan_sip.c:6480 determine_firstline_parts:
Bad request protocol Bad event

Also it will be great if anybody can tell where i can find the explanation
of all the warnig codes and error codes of asterisk if there is any.

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[asterisk-users] Sample configuration files for ATAs

[Rizwan Hisham]

Hi all,
I need some sample configuration files (in xml format) for some of my atas,
spa-2102, 1001, 2002, 3000. If anybody can provide these, i'll be very glad.


I have heard that some people can retrieve the configuration file from the
ata. I have all the above mentioned ata's, so if you can tell me how to take
out the conf files then it will also be very helpfull.

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[asterisk-users] Simple Question

[Rizwan Hisham]

Hi,
Does anyone know the purpose of "/n" attached at the end of the dial
command  below

Dial(Local/[EMAIL PROTECTED]/n )<

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[asterisk-users] Auto-congest time for sip peers

[Rizwan Hisham]

Hi all,
can anybody tell me how to make auto-congest time configurable(different)
for every sip peer. I mean if i want to dial a local number then i should be
able to set the autocongest time to 15000 mili seconds, but if i dial an
international number then i should be able to set the auto-congest time to
30,000 mili seconds. Can it be done in the dialplan? or should i jump into
the code?

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[asterisk-users] The switch statement in extensions.conf

[Rizwan Hisham]

Hi all,
i need to know how to use the switch statement in extensions.conf to throw
calls on a secondary asterisk server. I have been trying all day to be able
to make it work but its not working. It displays the following error:

pbx_find_extension: No such switch 'master:[EMAIL PROTECTED]'

I have read the help material on voip-info for connecting 2 asterisk servers
and followed the steps given there but no use.

Can anyone help me with this? Also need to know what the switch statement
does internally. i mean does the server which switches the call to another
server keeps any record of the call or it just transfers the call and
then all of the responsibility of the call is handled on the other server?

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