[asterisk-users] Dial application with M option

[ABBAS SHAKEEL]

Hello,

What i am trying to do is . Dail a number and ask if you wana talk to
XXX press 1 and if you dont wana talk press any other key.
For this purpose i am using  this
link
.
*I am using this option :- *
*M(**x**)*: Executes the macro (x) upon connect of the call (i.e. when the
called party answers). IMPORTANT - The CDR 'billsecs' field is set to zero
if the callee answers the call, but hangs up whilst the macro is still
running (if the callee answers and the macro finishes, 'billsecs' contains
the correct value).

*I am following this Example  :-*
*
*
*Example 2:* Dial macro
screen-record: Please record your name press pound when finished.
screen-from: You have a call from
screen-accept: Press 1 to accept this call or any other key to reject.

exten => 890,1,Wait(0.2)
exten => 890,n,Playback(screen-record)
exten => 890,n,SetVar(SCREEN_FILE=/tmp/${CALLERID(number)}-${EPOCH})
exten => 890,n,Record(${SCREEN_FILE}.gsm|6|25)
exten => 890,n,Dial(SIP/16|60|gM(screen^${SCREEN_FILE}))
exten => 890,n,Voicemail(1...@default)

[macro-screen]
exten => s,1,Wait(0.2)
exten => s,n,Playback(screen-from)
exten => s,n,Playback(${ARG1})
exten => s,n,Read(ACCEPT|screen-accept|1)
exten => s,n,GotoIf($[${ACCEPT} = 1 ] ?yes:no)
exten => s,n(yes),SetVar(MACRO_RESULT=CONTINUE)
exten => s,n(no),System(/bin/rm ${ARG1})
THE PROBLEM IS when the called party picks up the phone The call get
connected between the caller and callee. But I want it must ask the callee
if (s)he wana talk if press yes then connect else not. Please shed some
light on it
-- 
Kind Regards
Shakeel Abbas
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[asterisk-users] softphone @handheld

[Hans Witvliet]

Perhaps slightly O.T.

Does anybody know of (or even better, has experience with)
softphone clients on a blackberry?

Some friends of mine have those devices, but they can only use it for
data, voice has been disabled in their abo, so much of them carry they
business-BB, and their private GSM.

Rather "un-handy" afaics..

hw


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[asterisk-users] Fax throughput - Asterisk 1.6.1.9

[Cyprus VoIP]

Hello,

We are trying to send faxes by T.38 protocol to a remote SIP proxy from 
a local extension. The local extension sends the INVITE, Asterisk sends 
the call to the Proxy the call is connected with a regular audio codec. 
After a few seconds the remote proxy sends an INVITE with UDPTL and the 
Asterisk sends it to the local extension and it's accepted, but (here 
the problem starts) just after sending the OK with the proper SDP to the 
remote Proxy, the Asterisk initiates a new INVITE to the local extension 
and remote Proxy, with the normal audio codecs again.

We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the 
local extension and remote Proxy, but it still forces the call to go 
back to a voice call.

Any idea why it happens and how to debug it? We set verbose and debug to 
20, but no "internal" info is provided to get a clear understanding on 
Asterisk's "thoughts" during that process.

Thank you in advance for your assistance,

Andreas

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Re: [asterisk-users] dahdi_tool shows no alarms, but no line connected

[Tzafrir Cohen]

On Thu, Dec 03, 2009 at 01:13:13AM +0100, Roger Schreiter wrote:
> Hi,
> 
> I'm using Sangoma's wanpipe together with dahdi, all
> software downloaded today at most recent version.
> Hardware is Sangoma A104, a 4xE1 card.
> Installation went well.
> 
> Anyway, wanrouter status shows a different result than
> dahdi_tool or dahdi_scan.
> 
> I've just put a hardware loop on port 1. All the other
> ports are open.
> 
> wanrouter status shows the expected result:
> Device name | Protocol | Station | Status|
> wanpipe1| AFT TE1  | N/A | Connected |
> wanpipe2| AFT TE1  | N/A | Disconnected  |
> wanpipe3| AFT TE1  | N/A | Disconnected  |
> wanpipe4| AFT TE1  | N/A | Disconnected  |
> 
> However:
> # dahdi_scan 2
> [2]
> active=yes
> alarms=OK
> description=wanpipe2 card 1
> name=WPE1/1
> manufacturer=
> devicetype=
> location=
> basechan=1
> totchans=31
> irq=0
> type=digital-
> syncsrc=0
> lbo=0 db (CSU)/0-133 feet (DSX-1)
> coding_opts=HDB3
> framing_opts=CCS,CRC4
> coding=HDB3
> framing=CCS
> 
> 
> Why are the dahdi tools not reflecting the values
> by wanrouter?

What does the driver report?

head -n1 /proc/dahdi/*

If there are no alarms there, the wanpipe driver probably did not report
them to DAHDI.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Dial application with M option

[ABBAS SHAKEEL]

Any one have success with Dial M option, Can some one provide an example?

On Thu, Dec 3, 2009 at 12:57 PM, ABBAS SHAKEEL
wrote:

> Hello,
>
> What i am trying to do is . Dail a number and ask if you wana talk to
> XXX press 1 and if you dont wana talk press any other key.
> For this purpose i am using  this 
> link
> .
> *I am using this option :- *
> *M(**x**)*: Executes the macro (x) upon connect of the call (i.e. when the
> called party answers). IMPORTANT - The CDR 'billsecs' field is set to zero
> if the callee answers the call, but hangs up whilst the macro is still
> running (if the callee answers and the macro finishes, 'billsecs' contains
> the correct value).
>
> *I am following this Example  :-*
> *
> *
> *Example 2:* Dial macro
>  screen-record: Please record your name press pound when finished.
> screen-from: You have a call from
> screen-accept: Press 1 to accept this call or any other key to reject.
>
> exten => 890,1,Wait(0.2)
> exten => 890,n,Playback(screen-record)
> exten => 890,n,SetVar(SCREEN_FILE=/tmp/${CALLERID(number)}-${EPOCH})
> exten => 890,n,Record(${SCREEN_FILE}.gsm|6|25)
> exten => 890,n,Dial(SIP/16|60|gM(screen^${SCREEN_FILE}))
> exten => 890,n,Voicemail(1...@default)
>
> [macro-screen]
> exten => s,1,Wait(0.2)
> exten => s,n,Playback(screen-from)
> exten => s,n,Playback(${ARG1})
> exten => s,n,Read(ACCEPT|screen-accept|1)
> exten => s,n,GotoIf($[${ACCEPT} = 1 ] ?yes:no)
> exten => s,n(yes),SetVar(MACRO_RESULT=CONTINUE)
> exten => s,n(no),System(/bin/rm ${ARG1})
> THE PROBLEM IS when the called party picks up the phone The call get
> connected between the caller and callee. But I want it must ask the callee
> if (s)he wana talk if press yes then connect else not. Please shed some
> light on it
> --
> Kind Regards
> Shakeel Abbas
>
>


-- 
Best Regards
Shakeel Abbas
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Re: [asterisk-users] Feature Request: "SayNumberFiles"

[Olivier]

2009/12/3 Warren Selby 

> Why not do something with Background()?  i.e
>
> Playback(you-have)
> SayNumber(${numMessages})
> Playback(messages)
> Background(press-1-or-2)
>
> Then just be sure to record the audio files in the appropriate
> directory...
>

The benefit of using a single Read (instead of  using several Playback,
Background and SayNumber) is that  IVR users don't have to wait for the last
prompt to input their reply.
I can be frustrating for users to be sometimes unable to reply without
waiting, just because the audio message can't be pre-recorded (because it
includes some variable data).


>
>
>
> Thanks,
> --Warren Selby
>
> On Dec 3, 2009, at 12:39 AM, Olivier  wrote:
>
> > Hi,
> >
> > Currently, it seems impossible to use the output of SayNumber
> > application as an input to Read application.
> > So, if you want to develop an IVR with something like "You've got 23
> > messages. Type 1 to listen to the first one. Type 2 to leave", you
> > must parse this message into 3 pieces and want for the last one play
> > to start listening of user input :
> >   Playback ( "You've got")
> >   SayNumber (23)
> >   Read ("messages. Type 1 to listen to the first one. Type 2 to
> > leave", ...)
> >
> > Being to get the list of sound files that SayNumber would play would
> > be very convenient to build concatenate several one into one
> > temporary file. When played with Read app, user wouldn't have to
> > wait for the last part to type its answer :
> >
> >   myfilelist = SayNumberFiles(23)
> >   tempfile=concat( "You've got", myfilelist, "messages. Type 1 to
> > listen to the first one. Type 2 to leave")
> >   Read (tempfile, ...)
> >
> > Did I miss a workaround ?
> > What would you say about this feature request ?
> >
> > Regards
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] dahdi_tool shows no alarms, but no line connected

[Roger Schreiter]

Tzafrir Cohen schrieb:
> ...
> head -n1 /proc/dahdi/*

# head -n1 /proc/dahdi/*
==> /proc/dahdi/1 <==
Span 1: WPE1/0 "wanpipe1 card 0" (MASTER) HDB3/CCS/CRC4

==> /proc/dahdi/2 <==
Span 2: WPE1/1 "wanpipe2 card 1" HDB3/CCS/CRC4

==> /proc/dahdi/3 <==
Span 3: WPE1/2 "wanpipe3 card 2" HDB3/CCS/CRC4

==> /proc/dahdi/4 <==
Span 4: WPE1/3 "wanpipe4 card 3" HDB3/CCS/CRC4


> If there are no alarms there, the wanpipe driver probably did not report
> them to DAHDI.

Probably. But why? (When I turn of wanpipes, I can
see them disapear with dahdi_tool.)

How can I investigate the reason?

Roger.


-- 
Roger Schreiter
Spindelberg 11
D-74354 Besigheim
Tel.: +49 7143 36476

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Re: [asterisk-users] Dial application with M option

[ABBAS SHAKEEL]

Aah the Problem was i am working on 1.4 and in my mind and logic i was
writing code for 1.6.

The example works perfect


On Thu, Dec 3, 2009 at 3:00 PM, ABBAS SHAKEEL
wrote:

> Any one have success with Dial M option, Can some one provide an example?
>
>
> On Thu, Dec 3, 2009 at 12:57 PM, ABBAS SHAKEEL <
> shakeel.abbas@gmail.com> wrote:
>
>> Hello,
>>
>> What i am trying to do is . Dail a number and ask if you wana talk to
>> XXX press 1 and if you dont wana talk press any other key.
>> For this purpose i am using  this 
>> link
>> .
>> *I am using this option :- *
>> *M(**x**)*: Executes the macro (x) upon connect of the call (i.e. when
>> the called party answers). IMPORTANT - The CDR 'billsecs' field is set to
>> zero if the callee answers the call, but hangs up whilst the macro is still
>> running (if the callee answers and the macro finishes, 'billsecs' contains
>> the correct value).
>>
>> *I am following this Example  :-*
>> *
>> *
>> *Example 2:* Dial macro
>>  screen-record: Please record your name press pound when finished.
>> screen-from: You have a call from
>> screen-accept: Press 1 to accept this call or any other key to reject.
>>
>> exten => 890,1,Wait(0.2)
>> exten => 890,n,Playback(screen-record)
>> exten => 890,n,SetVar(SCREEN_FILE=/tmp/${CALLERID(number)}-${EPOCH})
>> exten => 890,n,Record(${SCREEN_FILE}.gsm|6|25)
>> exten => 890,n,Dial(SIP/16|60|gM(screen^${SCREEN_FILE}))
>> exten => 890,n,Voicemail(1...@default)
>>
>> [macro-screen]
>> exten => s,1,Wait(0.2)
>> exten => s,n,Playback(screen-from)
>> exten => s,n,Playback(${ARG1})
>> exten => s,n,Read(ACCEPT|screen-accept|1)
>> exten => s,n,GotoIf($[${ACCEPT} = 1 ] ?yes:no)
>> exten => s,n(yes),SetVar(MACRO_RESULT=CONTINUE)
>> exten => s,n(no),System(/bin/rm ${ARG1})
>> THE PROBLEM IS when the called party picks up the phone The call get
>> connected between the caller and callee. But I want it must ask the callee
>> if (s)he wana talk if press yes then connect else not. Please shed some
>> light on it
>> --
>> Kind Regards
>> Shakeel Abbas
>>
>>
>
>
> --
> Best Regards
> Shakeel Abbas
>
>


-- 
Best Regards
Shakeel Abbas
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[asterisk-users] AEL, 1.6, CUT and commas

[Olivier]

Hello,

How can you parse a comma separated list using function CUT and AEL ?

I've tried but it displays error message (though is seems to find the
correct value) :

STRING=101,102
VAL=${CUT(STRING,\,,1)};
NoOp(VAL is ${VAL});

Cheers
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Re: [asterisk-users] AEL, 1.6, CUT and commas [SOLVED]

[Olivier]

2009/12/3 Olivier 

> Hello,
>
> How can you parse a comma separated list using function CUT and AEL ?
>
> I've tried but it displays error message (though is seems to find the
> correct value) :
>
> STRING=101,102
> VAL=${CUT(STRING,\,,1)};
> NoOp(VAL is ${VAL});
>
> Cheers
>

Sorry for the noise but I mixed up with another mistake elsewhere in my
code.
The above example works !
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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

[Kevin P. Fleming]

Cyprus VoIP wrote:

> We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the 
> local extension and remote Proxy, but it still forces the call to go 
> back to a voice call.

Define 'internal extension'. Is this a T.38-capable device? If not,
Asterisk doesn't support TDM-to-T.38 FAX relay (yet). If it is, then the
path to resolving this problem is to collect a complete console log of
the failing call, including 'core set debug 10', 'core set verbose 10'
and 'sip set debug on' (and please ensure that all five logger levels
are enabled for the 'console' log channel in logger.conf).

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] ChanSpy gets stuck

[Joao Gomes Pereira]

Hello
I have a simple configuration to allow the admins listen to agent calls:

exten => _654,1,ChanSpy(Agent)  
exten => _654,2,Hangup()


The problem is... even when the agents hung up... it seems the channels 
remain active:

okavango*CLI> show channels
SIP/211-b3042018 6...@default:1Up  
ChanSpy(Agent)   
SIP/211-b3fbf768 6...@default:1Up  
ChanSpy(Agent)   
SIP/211-b4d88940 6...@default:1Up  
ChanSpy(Agent)   
SIP/211-b52e6498 6...@default:1Up  
ChanSpy(Agent)   
SIP/211-b4bbfcc0 6...@default:1Up  
ChanSpy(Agent)   
SIP/211-b4ba9f88 6...@default:1Up  
ChanSpy(Agent)   

(the agent is using extension 211)

I have more then 10 lines like these. Why do the ChanSpy calls dont hang up?
Thanks
Regards
Joao Pereira

-- 
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt


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[asterisk-users] Wi-Fi sip phones with auto provisioning

[Lefteris Zafiris]

Im looking for wifi sip phones that support auto provisioning and work
flawlessly with atserisk. Can anyone suggest me some models?

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Re: [asterisk-users] Wi-Fi sip phones with auto provisioning

[Fred Posner]

On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote:

> Im looking for wifi sip phones that support auto provisioning and work
> flawlessly with atserisk. Can anyone suggest me some models?
> 

Don't know of any wifi phone that works flawlessly whatsoever. Best to consider 
a DECT style phone.

---fred
http://qxork.com


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Re: [asterisk-users] Wi-Fi sip phones with auto provisioning

[Lefteris Zafiris]

Fred Posner wrote:
> On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote:
> 
>> Im looking for wifi sip phones that support auto provisioning and work
>> flawlessly with atserisk. Can anyone suggest me some models?
>>
> 
> Don't know of any wifi phone that works flawlessly whatsoever. Best to 
> consider a DECT style phone.
> 
> ---fred
> http://qxork.com
> 

Ok i can live with wifi phones that work *almost* flawlessly with
asterisk :D
DECT is not an option for now.


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Re: [asterisk-users] Feature Request: "SayNumberFiles"

[Danny Nicholas]

With a little familiarity of how SayNumber works, this isn't hard at all.
All that Saynumbers does is parse out the number into 1, 2 or 3 digits and
then plays files from /var/lib/asterisk/sounds/digits.  In the example the
OP posted, ${numMessages} had a value of 23, so 20.gsm and 3.gsm would be
played back.  To use the read function's single file "capability", you could
use sox to concatenate the files together, or just use the playback example
below or use an AGI like this:
exten => s,1,AGI(nomessages.agi|${nummessages})
exten => s,n,Background(${PART1}&${PART2}&${PART3}&${PART4})
exten => s,n,Read(faxno,beep,1,skip,1,5)  

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Thursday, December 03, 2009 12:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Feature Request: "SayNumberFiles"

Why not do something with Background()?  i.e

Playback(you-have)
SayNumber(${numMessages})
Playback(messages)
Background(press-1-or-2)

Then just be sure to record the audio files in the appropriate  
directory...




Thanks,
--Warren Selby

On Dec 3, 2009, at 12:39 AM, Olivier  wrote:

> Hi,
>
> Currently, it seems impossible to use the output of SayNumber  
> application as an input to Read application.
> So, if you want to develop an IVR with something like "You've got 23  
> messages. Type 1 to listen to the first one. Type 2 to leave", you  
> must parse this message into 3 pieces and want for the last one play  
> to start listening of user input :
>   Playback ( "You've got")
>   SayNumber (23)
>   Read ("messages. Type 1 to listen to the first one. Type 2 to  
> leave", ...)
>
> Being to get the list of sound files that SayNumber would play would  
> be very convenient to build concatenate several one into one  
> temporary file. When played with Read app, user wouldn't have to  
> wait for the last part to type its answer :
>
>   myfilelist = SayNumberFiles(23)
>   tempfile=concat( "You've got", myfilelist, "messages. Type 1 to  
> listen to the first one. Type 2 to leave")
>   Read (tempfile, ...)
>
> Did I miss a workaround ?
> What would you say about this feature request ?
>
> Regards
> ___
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>
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Re: [asterisk-users] Wi-Fi sip phones with auto provisioning

[Steve Howes]

On 3 Dec 2009, at 13:55, Fred Posner wrote:

>
> On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote:
>
>> Im looking for wifi sip phones that support auto provisioning and  
>> work
>> flawlessly with atserisk. Can anyone suggest me some models?
>>
>
> Don't know of any wifi phone that works flawlessly whatsoever. Best  
> to consider a DECT style phone.

If you do go Wifi, don't get a WIP310...

S

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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

[David Backeberg]

Cyprus VoIP wrote:
> We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the
> local extension and remote Proxy, but it still forces the call to go
> back to a voice call.

That's correct behavior if T.38 cannot autonegotiate.

What happens in the reverse direction, trying to send faxes from your
mysterious proxy to asterisk?

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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

[Cyprus VoIP]

>> We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the 
>> local extension and remote Proxy, but it still forces the call to go 
>> back to a voice call.
> 
> Define 'internal extension'. Is this a T.38-capable device? If not,
> Asterisk doesn't support TDM-to-T.38 FAX relay (yet). If it is, then the
> path to resolving this problem is to collect a complete console log of
> the failing call, including 'core set debug 10', 'core set verbose 10'
> and 'sip set debug on' (and please ensure that all five logger levels
> are enabled for the 'console' log channel in logger.conf).
> 

Hi Kevin,

Thank you for your answer. The 'internal extension' is indeed a T.38 
capable device that works perfectly when connected directly to the 
Proxy/ITSP.

As you said, the key to debugging/resolving this issue is the logger. I 
wasn't aware of this file. this is what I have there:
...
;debug => debug
console => notice,warning,error
;console => notice,warning,error,debug
messages => notice,warning,error
;full => notice,warning,error,debug,verbose
...

Should I change the "console..." line or uncomment the ";full..." line?

Regards,

Andreas

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Re: [asterisk-users] Wi-Fi sip phones with auto provisioning

[Mark Best]

The Ascom i75 isn't really an 'auto-provision' out of the box WiFi
phone, but it has a fairly painless USB cradle for programming. Works
well with Asterisk & DD-WRT.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: Thursday, December 03, 2009 06:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Wi-Fi sip phones with auto provisioning


On 3 Dec 2009, at 13:55, Fred Posner wrote:

>
> On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote:
>
>> Im looking for wifi sip phones that support auto provisioning and  
>> work
>> flawlessly with atserisk. Can anyone suggest me some models?
>>
>
> Don't know of any wifi phone that works flawlessly whatsoever. Best  
> to consider a DECT style phone.

If you do go Wifi, don't get a WIP310...

S

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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

[Kevin P. Fleming]

Cyprus VoIP wrote:

> Thank you for your answer. The 'internal extension' is indeed a T.38 
> capable device that works perfectly when connected directly to the 
> Proxy/ITSP.
> 
> As you said, the key to debugging/resolving this issue is the logger. I 
> wasn't aware of this file. this is what I have there:
> ...
> ;debug => debug
> console => notice,warning,error
> ;console => notice,warning,error,debug
> messages => notice,warning,error
> ;full => notice,warning,error,debug,verbose
> ...
> 
> Should I change the "console..." line or uncomment the ";full..." line?

Either one is fine; using 'full' is actually a bit better, because the
color highlighting done on the console sometimes makes console captures
hard to read.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Wi-Fi sip phones with auto provisioning

[Connor Spiess]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: Thursday, December 03, 2009 06:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Wi-Fi sip phones with auto provisioning


On 3 Dec 2009, at 13:55, Fred Posner wrote:

>>
>> On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote:
>>
>>> Im looking for wifi sip phones that support auto provisioning and
>>> work
>>> flawlessly with atserisk. Can anyone suggest me some models?
>>>
>>
>> Don't know of any wifi phone that works flawlessly whatsoever. Best
>> to consider a DECT style phone.
>
>The Ascom i75 isn't really an 'auto-provision' out of the box WiFi
>phone, but it has a fairly painless USB cradle for programming. Works
>well with Asterisk & DD-WRT.

You can purchase an IMS and PDM and use Ascom's Virtual SIM technology which 
lets for simple provisioning by just entering in the extension number you want 
the phone to be over the air. Or as Mark said you can use the desktop cradle 
which fairly painless process.

I have had good look with the Ascom's and use one everyday as my desk phone.
I have also used the Polycom/Spectralink phones and can provision via TFTP. I 
have only played with these a little bit here and there so I can't speak to 
overall reliability.

--Connor

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Re: [asterisk-users] Wi-Fi sip phones with auto provisioning

[Thorolf Godawa]

Hi,

> Im looking for wifi sip phones that support auto provisioning and work
> flawlessly with atserisk. Can anyone suggest me some models?
I have not yet tested it with provisioning because right now the
provisioning server does not support this model.

But I use a Snom 820 and 870 with an extra USB-Wifi-stick (ALLNET
ALL0233RP 300MBit) instead of ethernet and this works quite good.

Unfortunately I do not like the 870 at all, bad quality, bad
user-interface (touch screen), for me the 820 is better to use.

I also know that the Siemens OpenStage 60 SIP also has an USB-port, but
I have not checked if it will work with an USB-Wifi-stick too!
-- 

Chau y hasta luego,

Thorolf

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[asterisk-users] multiple sip trunks

[John Taylor]

I want to use an asterisk box to provide a voip service to a number of
separate companies.

I have a VOIP provider who I want to trunk with. As far as I can see
it there are 2 options
1. Have 1 SIP trunk to one account at the provider who gives me
multiple incoming numbers; this is less than optimal as the provider
does not provide the DID number in the sip header; I only get the
account number. I have the option to set "called line presentation"
but this will stop CLID

2. Have multiple sip trunks to multiple accounts at the provider. Is
this an advisable thing to do? I notice asterisk does not handle the
incoming context correctly (all incoming calls go to the last incoming
context defined in sip.conf), but I can extract the account called via
the EXTEN variable.

I would be looking at providing around 20 companies with accounts (all
very small), and would prefer option (2) to enable failover to a
number they specify.

Thanks for any light shed

John

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[asterisk-users] Repost: Working in useful examples... and freenum/e.164 dialing in extensions.conf.example

[Philip A. Prindeville]

I've recently decided to spend idle cycles while waiting for various Astlinux 
platform builds to complete on making the contents of asterisk/config a little 
more complete, a little more useful, a little more real-world...

I started looking at the possibility of taking JTodd's ISN Freenum cookbook 
example and updating it:

http://freenum.org/cookbook/#ASTERISK_CONFIG

but ran into some problems having to do with the following:  we use SIP 
handsets in house, and those operate in the redfish-solutions.com domain... but 
anonymous SIP (freenum) calls coming in off the internet would also be in the 
same domain.  I can't figure out how to separate calls in the same domain (but 
from different endpoints) in two different contexts.

And unfortunately, doc/ doesn't cover sip.conf, nor does the Asterisk Reference 
Manual cover it in much detail... except for a couple of related topics (shared 
line appearances and dahdi integration, I think).

So if anyone can contact me and work off-line on getting a reasonably useful 
and real-world applicable example working, I'll file a documentation defect and 
push for getting this checked in (as I did for a few other examples).

Thanks,

-Philip


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[asterisk-users] Source-IP on Asterisk DRBD/-HA-Cluster wrong

[Thorolf Godawa]

Hi all,

I installed a Linux-HA-cluster with DRBD and Asterisk 1.4 on it.

Actually it might work quite good, failover etc. works, even if this is
not a 0-downtime solution, because current calls are dropped and the
phones not are reachable until they reregister at the Asterisk.

It "might" work good, because my phones are not able to register at the
cluster-ip address, because the REGISTER arrives at the cluster-ip,
while Asterisk responds from the real host-ip of the active cluster-node.

The registration with the real ip works, but then I can't use any
failover solution.

I have already checked several solutions found at
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
and some others returned by searching the net, but nothing that helped
me out.

I see three possible solutions:
- is there a parameter in Asterisk to set the source-ip?
- patching Asterisk to include such a parameter?
- would be changing the primary ip-address to the "cluster"-ip and
  changing the "real" host-ip to an alias, a good solution?

Thanks for any hints in advance,
-- 

Chau y hasta luego,

Thorolf

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Re: [asterisk-users] Source-IP on Asterisk DRBD/-HA-Cluster wrong

[Scott L. Lykens]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Thorolf Godawa
> Sent: Thursday, December 03, 2009 2:49 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Source-IP on Asterisk DRBD/-HA-Cluster wrong


> Actually it might work quite good, failover etc. works, even if this
is not a 0-
> downtime solution, because current calls are dropped and the phones
not
> are reachable until they reregister at the Asterisk.

Apologize for not directly answering your questions, however, I'm
considering playing with Remus and Xen in the future to deal with high
availability without dropping calls.

See http://dsg.cs.ubc.ca/remus/ for some details.

I have no idea if it will work or what the implications are but I
noticed that in doing research for some other projects and made a note
of it to try in the future.

sl

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Re: [asterisk-users] Source-IP on Asterisk DRBD/-HA-Cluster wrong

[Alex Balashov]

Thorolf,

In the [general] section of sip.conf, set the 'bindaddr' parameter to 
the cluster IP.  If Asterisk is only bound to the floating interface, it 
will respond only from that source IP.

-- Alex

Thorolf Godawa wrote:

> Hi all,
> 
> I installed a Linux-HA-cluster with DRBD and Asterisk 1.4 on it.
> 
> Actually it might work quite good, failover etc. works, even if this is
> not a 0-downtime solution, because current calls are dropped and the
> phones not are reachable until they reregister at the Asterisk.
> 
> It "might" work good, because my phones are not able to register at the
> cluster-ip address, because the REGISTER arrives at the cluster-ip,
> while Asterisk responds from the real host-ip of the active cluster-node.
> 
> The registration with the real ip works, but then I can't use any
> failover solution.
> 
> I have already checked several solutions found at
> http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
> and some others returned by searching the net, but nothing that helped
> me out.
> 
> I see three possible solutions:
> - is there a parameter in Asterisk to set the source-ip?
> - patching Asterisk to include such a parameter?
> - would be changing the primary ip-address to the "cluster"-ip and
>   changing the "real" host-ip to an alias, a good solution?
> 
> Thanks for any hints in advance,


-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] multiple sip trunks

[Tim Nelson]

- "John Taylor"  wrote:
> I want to use an asterisk box to provide a voip service to a number
> of
> separate companies.
> 
> I have a VOIP provider who I want to trunk with. As far as I can see
> it there are 2 options
> 1. Have 1 SIP trunk to one account at the provider who gives me
> multiple incoming numbers; this is less than optimal as the provider
> does not provide the DID number in the sip header; I only get the
> account number. I have the option to set "called line presentation"
> but this will stop CLID
> 
> 2. Have multiple sip trunks to multiple accounts at the provider. Is
> this an advisable thing to do? I notice asterisk does not handle the
> incoming context correctly (all incoming calls go to the last
> incoming
> context defined in sip.conf), but I can extract the account called
> via
> the EXTEN variable.
> 
> I would be looking at providing around 20 companies with accounts
> (all
> very small), and would prefer option (2) to enable failover to a
> number they specify.
> 
> Thanks for any light shed
> 
> John
> 

Why not go with a real carrier that can send you proper DID and DNIS 
information for each call? Rather than trying to configure/code/etc around the 
problem with the ITSP, use an ITSP that does things correctly. There are many 
people here on asterisk-users that can recommend a proper ITSP. If you want 
pure business response, head over to asterisk-biz and ask there.

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

[Alex Balashov]

Set 'canreinvite=no' on all applicable peers?

Cyprus VoIP wrote:

> Hello,
> 
> We are trying to send faxes by T.38 protocol to a remote SIP proxy from 
> a local extension. The local extension sends the INVITE, Asterisk sends 
> the call to the Proxy the call is connected with a regular audio codec. 
> After a few seconds the remote proxy sends an INVITE with UDPTL and the 
> Asterisk sends it to the local extension and it's accepted, but (here 
> the problem starts) just after sending the OK with the proper SDP to the 
> remote Proxy, the Asterisk initiates a new INVITE to the local extension 
> and remote Proxy, with the normal audio codecs again.
> 
> We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the 
> local extension and remote Proxy, but it still forces the call to go 
> back to a voice call.
> 
> Any idea why it happens and how to debug it? We set verbose and debug to 
> 20, but no "internal" info is provided to get a clear understanding on 
> Asterisk's "thoughts" during that process.
> 
> Thank you in advance for your assistance,
> 
> Andreas
> 
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-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] Source-IP on Asterisk DRBD/-HA-Cluster wrong

[Matt Riddell]

On 4/12/09 9:28 AM, Scott L. Lykens wrote:
> Apologize for not directly answering your questions, however, I'm
> considering playing with Remus and Xen in the future to deal with high
> availability without dropping calls.
>
> See http://dsg.cs.ubc.ca/remus/ for some details.
>
> I have no idea if it will work or what the implications are but I
> noticed that in doing research for some other projects and made a note
> of it to try in the future.

Yeah, I was looking at that too - haven't had much time to work on it 
further, but if it can handle running Asterisk cleanly it looks like a 
pretty nice solution.

I'm just not 100% convinced that it will work in a real time environment 
rather than for hosting web sites.

I'm sure that it would be perfect if you wanted failover web sites 
without any downtime, but wonder how it would work with Asterisk.

Post your progress as you move through it :)


-- 
Cheers,

Matt Riddell
Director
___

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http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

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Re: [asterisk-users] Source-IP on Asterisk DRBD/-HA-Cluster wrong

[Fred Posner]

On Dec 3, 2009, at 5:05 PM, Matt Riddell wrote:

> On 4/12/09 9:28 AM, Scott L. Lykens wrote:
>> Apologize for not directly answering your questions, however, I'm
>> considering playing with Remus and Xen in the future to deal with high
>> availability without dropping calls.
>> 
>> See http://dsg.cs.ubc.ca/remus/ for some details.
>> 
>> I have no idea if it will work or what the implications are but I
>> noticed that in doing research for some other projects and made a note
>> of it to try in the future.
> 
> Yeah, I was looking at that too - haven't had much time to work on it 
> further, but if it can handle running Asterisk cleanly it looks like a 
> pretty nice solution.
> 
> I'm just not 100% convinced that it will work in a real time environment 
> rather than for hosting web sites.
> 
> I'm sure that it would be perfect if you wanted failover web sites 
> without any downtime, but wonder how it would work with Asterisk.
> 
> Post your progress as you move through it :)
> 

If you're using just SIP to SIP, a better option would be a pure sip proxy, ala 
Kamailio/SER, etc. They can survive a failover without a drop. If you're using 
Asterisk during the call for media, recording, etc., I couldn't think of a 
solution where that would not result in the calls on the specific instance 
ending.

---fred
http://qxork.com






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Re: [asterisk-users] Source-IP on Asterisk DRBD/-HA-Cluster wrong

[Alex Balashov]

Fred Posner wrote:

> If you're using just SIP to SIP, a better option would be a pure sip proxy, 
> ala Kamailio/SER, etc. They can survive a failover without a drop. 

Agreed.  Even if using transaction-stateful relay mode, as long as a 
dialog is nailed up, sequential in-dialog messages (re-INVITEs, BYEs, 
etc.) can be routed based on the Route header even if the runtime 
transaction state has been lost.

-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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[asterisk-users] only the first ResetCDR works after upgrade to 1.6

[Andrew Witt]

Hello -

I am upgrading from asterisk v1.2 to v1.6 and I am seeing a problem with
recording CDRs using MySQL.  Unlike all of the other postings and web
pages I have found on this issue, my installation successfully stores
the -first- CDR, but nothing after that.

As background info, I will note that I don't use CDRs for billing, but
more in a logging fashion, to record how a given call branches through
the dialplan, the selections the caller makes, info retrieved from other
sources, etc.

The cdr_addon_mysql module loads; I have a working cdr_mysql.conf; MySQL
is running; etc., etc., etc.  The fact that the first time ResetCDR is
called it writes out a CDR shows that all the right pieces are in place,
so it isn't any of the basic installation, set-up and config issues.

By way of debugging, I pulled cdr_addon_mysql.c out of SVN (from
asterisk-addons/branches/1.6.2) and added a bunch of debug assertions
amidst the code that INSERTs the CDR record into the db.  I see in my
asterisk logs and on the asterisk console all of the expected assertions,
between the first "Executing ResetCDR" and the "Inserting a CDR record."

There are further "Executing ResetCDR" lines in the debug output, but
-no- further log lines from cdr_addon_mysql at all.

My dialplan is also stored in MySQL, and the debug output shows that
the connection from asterisk to MySQL remains up and functional, so I'm
fairly certain it is not a dropped MySQL connection.

It looks to me as if ResetCDR simply is not calling mysql_log( ) after
the first time ResetCDR is called.

Is this a known issue?  Has anyone else seen this behavior?  Does anyone
have a solution or at least a work-around?

Also, a number of other subtle but important things changed between v1.2
and v1.6 -- do I need to change my basic CDR logging strategy, too?
Currently, I have what amounts to a subroutine in the dialplan which
calls "Set(CDR(userfield)=" and then "ResetCDR(w)".  From what I have
read, this approach should still work under v1.6.

Many thanks in advance.  Asterisk rocks, and has served me well and been
very stable as my company's 24x7 payment IVR for nearly four years.  This
single issue is holding up my upgrade from v1.2 to v1.6, and I anticipate
many more solid years on 1.6 once this is resolved.  Thanks again.

-- 
Andrew Witt
Sr. Software Systems Engineer
revol wireless
216-525-1195 phone
216-240-1991 wireless
216-525-1112 fax
andrew.w...@revol.com
www.revol.com

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[asterisk-users] queue_variables() function

[Olivier]

Hello,

Has someone successfully used this QUEUE_VARIABLES() function (in 1.6.2-rc7)
?
I tried to use it as I'm using SIPPEER() but without success.

A previous question about it remainded unanswered (
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466).

Regards
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Re: [asterisk-users] Feature Request: "SayNumberFiles"

[Olivier]

2009/12/3 Danny Nicholas 

> With a little familiarity of how SayNumber works, this isn't hard at all.
> All that Saynumbers does is parse out the number into 1, 2 or 3 digits and
> then plays files from /var/lib/asterisk/sounds/digits.  In the example the
> OP posted, ${numMessages} had a value of 23, so 20.gsm and 3.gsm would be
> played back.


Doesn't that depend on language specifics (though I agree that for most,
using a single native language is enough) ?


>  To use the read function's single file "capability", you could
> use sox to concatenate the files together, or just use the playback example
> below or use an AGI like this:
> exten => s,1,AGI(nomessages.agi|${nummessages})
> exten => s,n,Background(${PART1}&${PART2}&${PART3}&${PART4})
> exten => s,n,Read(faxno,beep,1,skip,1,5)
>

You're right : it seems an interesting work around.


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
> Sent: Thursday, December 03, 2009 12:56 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Feature Request: "SayNumberFiles"
>
> Why not do something with Background()?  i.e
>
> Playback(you-have)
> SayNumber(${numMessages})
> Playback(messages)
> Background(press-1-or-2)
>
> Then just be sure to record the audio files in the appropriate
> directory...
>
>
>
>
> Thanks,
> --Warren Selby
>
> On Dec 3, 2009, at 12:39 AM, Olivier  wrote:
>
> > Hi,
> >
> > Currently, it seems impossible to use the output of SayNumber
> > application as an input to Read application.
> > So, if you want to develop an IVR with something like "You've got 23
> > messages. Type 1 to listen to the first one. Type 2 to leave", you
> > must parse this message into 3 pieces and want for the last one play
> > to start listening of user input :
> >   Playback ( "You've got")
> >   SayNumber (23)
> >   Read ("messages. Type 1 to listen to the first one. Type 2 to
> > leave", ...)
> >
> > Being to get the list of sound files that SayNumber would play would
> > be very convenient to build concatenate several one into one
> > temporary file. When played with Read app, user wouldn't have to
> > wait for the last part to type its answer :
> >
> >   myfilelist = SayNumberFiles(23)
> >   tempfile=concat( "You've got", myfilelist, "messages. Type 1 to
> > listen to the first one. Type 2 to leave")
> >   Read (tempfile, ...)
> >
> > Did I miss a workaround ?
> > What would you say about this feature request ?
> >
> > Regards
> > ___
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[asterisk-users] Multiple Channel Variables with AMI Originate

[Matthew Edmondson]

Hi guys I seem to be having a problem, I don't know if it's a bug or whether
I'm just doing it incorrectly.

I want to set about 3 channel variables when I originate a call via AMI.

All the documentation I have found says to do it like this:

Variable: variable1=value|variable2=value|variable3=value


However when I do this it runs them all together and I end up with:

variable1 = "value|variable2=value|variable3=value"


Instead of:

variable1 = "value"
variable2 = "value"
variable3 = "value"


So I think the delimiters are not working.

I tried just adding multiple "Variable:" lines one for each, but it only
sets the first one and ignores the rest.

Any help with this would be greatly appreciated.

Thanks! --Matt
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Re: [asterisk-users] Multiple Channel Variables with AMI Originate

[Jim Dickenson]

I do this:

Action: Originate
Channel: Local/dial_num...@cfmc_cdi_private
Exten: queue_answer
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=CallAndQueue
Variable: CfMC_QueueToUse=tqe
Variable: CfMC_AgentToUse=1001
Variable: CfMC_DialInfo=SIP/GXP280_18
Variable: CfMC_RingTimeout=30
Variable: CfMC_DoAMD=No
ActionID: CallAndQueue
Async: true


and things work as expected. This is in recent versions of 1.4 and 1.6.0.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Dec 3, 2009, at 4:22 PM, Matthew Edmondson wrote:

> 
> Hi guys I seem to be having a problem, I don't know if it's a bug or whether 
> I'm just doing it incorrectly.
> 
> I want to set about 3 channel variables when I originate a call via AMI.
> 
> All the documentation I have found says to do it like this:
> 
> Variable: variable1=value|variable2=value|variable3=value
> 
> 
> However when I do this it runs them all together and I end up with:
> 
> variable1 = "value|variable2=value|variable3=value"
> 
> 
> Instead of:
> 
> variable1 = "value"
> variable2 = "value"
> variable3 = "value"
> 
> 
> So I think the delimiters are not working.
> 
> I tried just adding multiple "Variable:" lines one for each, but it only sets 
> the first one and ignores the rest.
> 
> Any help with this would be greatly appreciated.
> 
> Thanks! --Matt
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Re: [asterisk-users] Multiple Channel Variables with AMI Originate

[Matthew Edmondson]

Thanks Jim.

We're using the 1.6.2 rc's. I'll try it on 1.6.0 and see if it works. If it
does then I guess its a bug in the new releases.

Problem is we need features of the 1.6.2.

On Fri, Dec 4, 2009 at 11:03 AM, Jim Dickenson  wrote:

> I do this:
>
> Action: Originate
> Channel: Local/dial_num...@cfmc_cdi_private
> Exten: queue_answer
> Context: cfmc_cdi_private
> Priority: 1
> Variable: CfMC_ActionID=CallAndQueue
> Variable: CfMC_QueueToUse=tqe
> Variable: CfMC_AgentToUse=1001
> Variable: CfMC_DialInfo=SIP/GXP280_18
> Variable: CfMC_RingTimeout=30
> Variable: CfMC_DoAMD=No
> ActionID: CallAndQueue
> Async: true
>
>
> and things work as expected. This is in recent versions of 1.4 and 1.6.0.
> --
> Jim Dickenson
> mailto:dicken...@cfmc.com 
>
> CfMC
> http://www.cfmc.com/
>
>
>
> On Dec 3, 2009, at 4:22 PM, Matthew Edmondson wrote:
>
>
> Hi guys I seem to be having a problem, I don't know if it's a bug or
> whether I'm just doing it incorrectly.
>
> I want to set about 3 channel variables when I originate a call via AMI.
>
> All the documentation I have found says to do it like this:
>
> Variable: variable1=value|variable2=value|variable3=value
>
>
> However when I do this it runs them all together and I end up with:
>
> variable1 = "value|variable2=value|variable3=value"
>
>
> Instead of:
>
> variable1 = "value"
> variable2 = "value"
> variable3 = "value"
>
>
> So I think the delimiters are not working.
>
> I tried just adding multiple "Variable:" lines one for each, but it only
> sets the first one and ignores the rest.
>
> Any help with this would be greatly appreciated.
>
> Thanks! --Matt
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>
>
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Re: [asterisk-users] Source-IP on Asterisk DRBD/-HA-Cluster wrong

[Thorolf Godawa]

Hi Alex,

> In the [general] section of sip.conf, set the 'bindaddr' parameter to
> the cluster IP.  If Asterisk is only bound to the floating interface,
yeah, that's it :-)

I have not tested failover right now, but registring of the phones now
works!

Thanks a lot,
-- 

Chau y hasta luego,

Thorolf

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Re: [asterisk-users] Source-IP on Asterisk DRBD/-HA-Cluster wrong

[Thorolf Godawa]

Hi Scott,

> Apologize for not directly answering your questions, however, I'm
> considering playing with Remus and Xen in the future to deal with
> high availability without dropping calls.
thanks a lot for the link, it looks quite interesting.

Unfortunately I think (and this is also my experience), a virtualized
Asterisk server will not work on higher load and might loose
UDP-VoIP-pakets what will result in a bad voice quality!
-- 

Chau y hasta luego,

Thorolf

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Re: [asterisk-users] Source-IP on Asterisk DRBD/-HA-Cluster wrong

[Alex Balashov]

Thorolf Godawa wrote:

> Unfortunately I think (and this is also my experience), a virtualized
> Asterisk server will not work on higher load and might loose
> UDP-VoIP-pakets what will result in a bad voice quality!

Much has been said of this topic.  In general, you are correct;  the 
effects of mostly userspace scheduling on timing tolerance and system 
responsiveness in a classical virtualisation scenario can lead to more 
jitter and stochastic variance in packet forwarding, even if not 
necessarily packet loss per se.

However, virtualisation technology is evolving in directions that make 
it increasingly "natively" bound to the underlying hardware - i.e. 
paravirtualisation.  Running Asterisk in a VZ-style VPS is very 
different than running it in a Xen VM slice, for example, because in 
the former case it is basically a glorified chroot jail;  there is a 
shared underlying kernel and simulated process space isolation.  In 
the latter, an actual virtual machine is run, although it is still 
native-bound to some degree in that it is not a pure 
userspace-scheduled process.

So, the answer is somewhat qualified;  it depends on what is meant by 
"generalisation."

-- Alex

-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] Multiple Channel Variables with AMI Originate

[Tilghman Lesher]

On Thursday 03 December 2009 18:22:19 Matthew Edmondson wrote:
> Hi guys I seem to be having a problem, I don't know if it's a bug or
> whether I'm just doing it incorrectly.
>
> I want to set about 3 channel variables when I originate a call via AMI.
>
> All the documentation I have found says to do it like this:
>
> Variable: variable1=value|variable2=value|variable3=value

With the change in application delimiter came this change, as it uses the
application delimiter parsing code.  Therefore, the variable delimiter is a
comma, not the pipe.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] chan_sip Error

[DHAVAL INDRODIYA]

hello
all,

i found this error on asterisk CLI while try to play file on SIP channel.
scenario is, manager send a command in meetme and meetme will play file to
user who connected via SIP channel.


chan_sip.c:5041 sip_write: Asked to transmit frame type 64, while native
formats is 0x2 (gsm)(2) read/write = 0x40 (slin)(64)/0x2 (gsm)(2)

can anybody know about this
what i m missing in configuration

regards
Dhaval
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[asterisk-users] hey please help me my 3rd email of how to change From fileld username in sip packet

[Masood Ahmed]

hy
Hope everyone is fine, I have one issue coming in asterisk , What i am doing
is i am generating a callback if some one calls at a specif access number on
asterisk,

Asterisk sends a busy signal to the calling party that he received a request
from party and then sends the call back to the person from where asterisk
received a request but in From field as you can see below astrisk is sending
the calling ID as "asterisk" and username same ,

What i want is that it should forward some CLI in From Field ,


I have done my best effort but still not resolved i am adding a callerid in
script still same please help me if some one can


 IP1:5060 -> IP2:5060
 INVITE sip:0423347871...@ip2:5060 SIP/2.0..Via: SIP/2.0/UDP
IP1:5060;branch=z9hG4bK-
966123148--16781
 75694--693700493-4-..Via: SIP/2.0/UDP
IP1:5065;branch=z9hG4bK00749b6d;rport..Call-Id: 4f8a33b207cd65f060b083b57
 804d...@ip1..to <804d...@117.20.20.234..to>: ..
*
*From: "asterisk";tag=as0cae0b**

see the last part this is what that i want to change here in from it should
be some CLI

thanks
Masood
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Re: [asterisk-users] hey please help me my 3rd email of how to change From fileld username in sip packet

[Hakan C]

exten => 111,1,Set(CallerID(num)=123456)

On Fri, Dec 4, 2009 at 8:32 AM, Masood Ahmed  wrote:

> hy
> Hope everyone is fine, I have one issue coming in asterisk , What i am
> doing
> is i am generating a callback if some one calls at a specif access number
> on
> asterisk,
>
> Asterisk sends a busy signal to the calling party that he received a
> request
> from party and then sends the call back to the person from where asterisk
> received a request but in From field as you can see below astrisk is
> sending
> the calling ID as "asterisk" and username same ,
>
> What i want is that it should forward some CLI in From Field ,
>
>
> I have done my best effort but still not resolved i am adding a callerid in
> script still same please help me if some one can
>
>
>  IP1:5060 -> IP2:5060
>  INVITE sip:0423347871...@ip2:5060 SIP/2.0..Via: SIP/2.0/UDP
> IP1:5060;branch=z9hG4bK-
> 966123148--16781
>  75694--693700493-4-..Via: SIP/2.0/UDP
> IP1:5065;branch=z9hG4bK00749b6d;rport..Call-Id: 4f8a33b207cd65f060b083b57
>  804d...@ip1..to <804d...@117.20.20.234..to>: ..
> *
> *From: "asterisk";tag=as0cae0b**
>
> see the last part this is what that i want to change here in from it should
> be some CLI
>
> thanks
> Masood
>
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