Re: [asterisk-users] get start-time of all active calls

[Sammy Govind]

Hi,
I think you need to use the command "sip show channel "
Regards,
Sammy

On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar wrote:

>  Hello,
>
> asterisk version 1.6.2.7
>
> I want to get the start time of all active calls from console, could you
> please let me know the best way to get it.
>
> thanks,
> Kamlesh
>
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> _
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[asterisk-users] get start-time of all active calls

[Kamlesh Kumar]

Hello,
 
asterisk version 1.6.2.7
 
I want to get the start time of all active calls from console, could you please 
let me know the best way to get it.
 
thanks,
Kamlesh   --
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Re: [asterisk-users] Realtime Registration

[Edwin Lam]

On 12/10/11 9:54 PM, Takehiro Matsushima wrote:


I'd configured realtime registration, but configuration was not applied when I
changed a row of sippeers table.
To apply, 'sip reload' was needed (in Asterisk 1.8.0).


or you can 'sip prune realtime '



(On 12/08/2011 03:42), Andrew O. Zhukov wrote:

No secrets :)

SELECT * FROM sippeers WHERE name = '105680' AND host = 'dynamic'

"name"|"type"|"username"|"secret"|"fromuser"|"fromdomain"|"nat"|"context"|"canreinvite"|"disallow"|"allow"|"host"|"insecure"|"port"|"ipaddr"|"outboundproxy"

"105680"|"peer"|"testbutton2"|"XXX"|""|"button.ipshka.com:5060"|"no"|"button"|"no"|"all"|"speex;ulaw;alaw;g729;ilbc;g726;g726aal2;slin;lpc10;adpcm;g723"|"dynamic"|"port,invite"|"5060
"|""|"ipshka.com"


On 12/07/2011 08:04 PM, Jonathan Rose wrote:

[Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer
'105680' is trying to register, but not configured as host=dynamic

Going on this, I'd say you probably tried to specify the host with a
static IP address or a host name. If that's the case, you can't
register, because that would be against the whole point of
registering in the first place.

You should probably post the DB entry for this peer to this thread
to make things simpler... if it doesn't contain sensitive data. Of
course, you can censor that out too.

- Original Message -
From: "Andrew O. Zhukov"
To: asterisk-users@lists.digium.com
Sent: Wednesday, December 7, 2011 11:56:20 AM
Subject: [asterisk-users] Realtime Registration

[Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:821 pgsql_reconnect:
Postgresql RealTime: Everything is fine.
[Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:201 realtime_pgsql:
Postgresql RealTime: Result=0x12e4edd0 Query: SELECT * FROM sippeers
WHERE name = '105680' AND host = 'dynamic'
[Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:209 realtime_pgsql:
Postgresql RealTime: Found 1 rows.
[Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer
'105680' is trying to register, but not configured as host=dynamic

Any suggestions???


Asterisk 1.4.42




--
Edwin Lam 
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20


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[asterisk-users] [hint state][BLF] Asterisk 1.8.7 does not send RINGING notifications, even with notifyringing=yes

[Niccolò Belli]

Hi,
I set verbose to 3, but I do not see any RINGING notification in the 
CLI. On the contrary, when the phone goes UNREACHABLE I get:


[Dec 13 21:10:06] NOTICE[9988]: chan_sip.c:25533 sip_poke_noanswer: Peer 
'152' is now UNREACHABLE! Last qualify: 130

== Extension Changed 152[blf] new state Unavailable for Notify User 154
[Dec 13 21:11:08] NOTICE[9988]: chan_sip.c:20196 
handle_response_peerpoke: Peer '152' is now Reachable. (528ms / 2000ms)

== Extension Changed 152[blf] new state Idle for Notify User 154

If I manually set the state to RINGING using 
Set(DEVICE_STATE(SIP/152)=RINGING) I get:


Extension Changed 152[blf] new state Ringing for Notify User 154

but otherwise I don't get any ringing notification.

Cheers,
Niccolò

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[asterisk-users] Login agents on asterisk startup according to hints state

[Niccolò Belli]

Hi,
I did map a key in each phone to add it to the incoming call queue 
(using AddQueueMember). It also updates a custom hint state for the Busy 
Lamp Field (BLF).
When I restart asterisk it keeps the previous Hint states (I don't know 
how, but it does), but obviously the phone is no longer a member of the 
queue. Is there a way to add phones to the incoming queue on asterisk 
startup according to the hint state?


Thanks,
Niccolò

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Re: [asterisk-users] AEL x LUA

[Matthew Nicholson]

On Tue, 2011-12-13 at 16:00 -0200, Antonio Modesto wrote:
> Hello,
> 
> I would like to receive some suggestions about dialplans written
> in lua, actually my dialplan is written in ael, but i'm having some
> problems with it. I noticed that asterisk just translates ael to the
> old extension language, does it do the same with lua?
> 

No, it does not translate the lua file into dialplan. The lua code is
executed as is.

-- 
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Digium, Inc. | Software Developer


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Re: [asterisk-users] Problem with Atxfer for the calling party

[Antonio Modesto]

On Tue, 2011-12-13 at 16:35 -0200, Roberto Linck wrote:

> Hi Antonio,
> 
> 
> 
> I'd never had used extensions.ael but in extensions.conf, using Macro
> I always set '__TRANSFER_CONTEXT' to the same context of exten and it
> works well.


Thanks, it worked, the MACRO_CONTEXT variable was empty, I've set the
value to my extensions context and it worked fine.


Thanks.

> 
> 
> 2011/12/13 Antonio Modesto 
> 
> Hello everybody,
> 
> I found that if i write my macro in the extensions.conf
> (not in ael), the atxfer works well, the problem is that ael
> uses gosub instead of the Macro() application, which doesn't
> change the current context. Does anybody know if i can do
> anything to solve this? I know if i rewrite all my macros in
> the common way, it will work, but that's a lot of coding for
> me.
> 
> 
> 
> 
> On Mon, 2011-12-12 at 08:57 -0200, Antonio Modesto wrote:
> 
> > Nothing?
> > 
> > 
> > On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote:
> > 
> > > 
> > > 
> > > 
> > > 
> > > Hi There,
> > > 
> > > I'm still having this problem, Does somebody  know
> > > what can be happening?
> > > 
> > > 
> > > Regards.
> > > 
> > > On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote:
> > > 
> > > > Hello,
> > > > 
> > > > The exten is the parameter passed to the macro,
> > > > which contains the sip device name. I'll change the name
> > > > to another less confusing.
> > > > 
> > > > * Alexandre, também sou brasileiro hehe, notei que você
> > > > já escreveu um livro sobre asterisk, será que você
> > > > poderia me ajudar com esse problema? Já tem alguns dias
> > > > que estou na luta aqui hehe.
> > > > 
> > > > On Fri, 2011-11-11 at 08:45 -0200, Alexandre Keller
> > > > wrote:
> > > > 
> > > > > You're using ${exten} inside your macro, you should
> > > > > use ${EXTEN}.
> > > > > -- 
> > > > > Atenciosamente,
> > > > > 
> > > > > ALEXANDRE KELLER
> > > > > 
> > > > > 
> > > > > http://twitter.com/alexandrekeller
> > > > > http://www.facebook.com/alexandre.keller.BR
> > > > > 
> > > > > "Dinheiro é a consequência de um trabalho bem feito e
> > > > > não o motivo para se fazer um bom trabalho."
> > > > > 
> > > > > 
> > > > > P Antes de imprimir pense em seu compromisso com
> > > > > o Meio Ambiente.
> > > > > 
> > > > > On 11/11/2011, at 08:38, Antonio Modesto wrote:
> > > > > 
> > > > > 
> > > > > > On Mon, 2011-11-07 at 09:12 -0600, Danny Nicholas
> > > > > > wrote:
> > > > > > 
> > > > > > > It can have to do with either the telephones dial
> > > > > > > plan or the context in the Asterisk dial plan
> > > > > > > combined with your features.conf settings.
> > > > > > 
> > > > > > 
> > > > > > I noticed that my problem occurs when i use a macro
> > > > > > to dial sip devices, my dialplan is like this:
> > > > > > 
> > > > > > - Each sip device has its own context
> > > > > > - This context includes the outgoing call contexts
> > > > > > that this extension can use for making calls and
> > > > > > includes a context called "ramais", which has the
> > > > > > dial plan to call another extensions, it uses a
> > > > > > macro to do this.
> > > > > > 
> > > > > > Here is the configuration for my extension
> > > > > > "modesto" :
> > > > > > 
> > > > > > # sip.conf
> > > > > > [modesto](default_extension)
> > > > > > username=modesto
> > > > > > context=modesto
> > > > > > callerid="modesto" <106>
> > > > > > callgroup=4
> > > > > > pickupgroup=4
> > > > > > 
> > > > > > # Default extension template
> > > > > > type=friend
> > > > > > dtmfmode=auto
> > > > > > host=dynamic
> > > > > > disallow=all
> > > > > > allow=ulaw
> > > > > > allow=alaw
> > > > > > deny=0.0.0.0/0.0.0.0
> > > > > > permit=192.168.1.0/255.255.255.0
> > > > > > canreinvite=yes
> > > > > > qualify=no
> > > > > > callcounter=yes
> > > > > > 
> > > > > > 
> > > > > > # context for SIP/modesto
> > > > > > context modesto {
> > > > > > includes {
> > > > > > vivo;
> > > > > > tim;
> > > > > > oi;
> > > > > > claro;
>

Re: [asterisk-users] Problem with Atxfer for the calling party

[Antonio Modesto]

On Tue, 2011-12-13 at 16:35 -0200, Roberto Linck wrote:

> Hi Antonio,
> 
> 
> 
> I'd never had used extensions.ael but in extensions.conf, using Macro
> I always set '__TRANSFER_CONTEXT' to the same context of exten and it
> works well.


Hello,

I'm already doing this, this line is inside my macro:

Set(__TRANSFER_CONTEXT=${MACRO_CONTEXT});

Is it right?

Thanks.


> 
> 
> 2011/12/13 Antonio Modesto 
> 
> Hello everybody,
> 
> I found that if i write my macro in the extensions.conf
> (not in ael), the atxfer works well, the problem is that ael
> uses gosub instead of the Macro() application, which doesn't
> change the current context. Does anybody know if i can do
> anything to solve this? I know if i rewrite all my macros in
> the common way, it will work, but that's a lot of coding for
> me.
> 
> 
> 
> 
> On Mon, 2011-12-12 at 08:57 -0200, Antonio Modesto wrote:
> 
> > Nothing?
> > 
> > 
> > On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote:
> > 
> > > 
> > > 
> > > 
> > > 
> > > Hi There,
> > > 
> > > I'm still having this problem, Does somebody  know
> > > what can be happening?
> > > 
> > > 
> > > Regards.
> > > 
> > > On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote:
> > > 
> > > > Hello,
> > > > 
> > > > The exten is the parameter passed to the macro,
> > > > which contains the sip device name. I'll change the name
> > > > to another less confusing.
> > > > 
> > > > * Alexandre, também sou brasileiro hehe, notei que você
> > > > já escreveu um livro sobre asterisk, será que você
> > > > poderia me ajudar com esse problema? Já tem alguns dias
> > > > que estou na luta aqui hehe.
> > > > 
> > > > On Fri, 2011-11-11 at 08:45 -0200, Alexandre Keller
> > > > wrote:
> > > > 
> > > > > You're using ${exten} inside your macro, you should
> > > > > use ${EXTEN}.
> > > > > -- 
> > > > > Atenciosamente,
> > > > > 
> > > > > ALEXANDRE KELLER
> > > > > 
> > > > > 
> > > > > http://twitter.com/alexandrekeller
> > > > > http://www.facebook.com/alexandre.keller.BR
> > > > > 
> > > > > "Dinheiro é a consequência de um trabalho bem feito e
> > > > > não o motivo para se fazer um bom trabalho."
> > > > > 
> > > > > 
> > > > > P Antes de imprimir pense em seu compromisso com
> > > > > o Meio Ambiente.
> > > > > 
> > > > > On 11/11/2011, at 08:38, Antonio Modesto wrote:
> > > > > 
> > > > > 
> > > > > > On Mon, 2011-11-07 at 09:12 -0600, Danny Nicholas
> > > > > > wrote:
> > > > > > 
> > > > > > > It can have to do with either the telephones dial
> > > > > > > plan or the context in the Asterisk dial plan
> > > > > > > combined with your features.conf settings.
> > > > > > 
> > > > > > 
> > > > > > I noticed that my problem occurs when i use a macro
> > > > > > to dial sip devices, my dialplan is like this:
> > > > > > 
> > > > > > - Each sip device has its own context
> > > > > > - This context includes the outgoing call contexts
> > > > > > that this extension can use for making calls and
> > > > > > includes a context called "ramais", which has the
> > > > > > dial plan to call another extensions, it uses a
> > > > > > macro to do this.
> > > > > > 
> > > > > > Here is the configuration for my extension
> > > > > > "modesto" :
> > > > > > 
> > > > > > # sip.conf
> > > > > > [modesto](default_extension)
> > > > > > username=modesto
> > > > > > context=modesto
> > > > > > callerid="modesto" <106>
> > > > > > callgroup=4
> > > > > > pickupgroup=4
> > > > > > 
> > > > > > # Default extension template
> > > > > > type=friend
> > > > > > dtmfmode=auto
> > > > > > host=dynamic
> > > > > > disallow=all
> > > > > > allow=ulaw
> > > > > > allow=alaw
> > > > > > deny=0.0.0.0/0.0.0.0
> > > > > > permit=192.168.1.0/255.255.255.0
> > > > > > canreinvite=yes
> > > > > > qualify=no
> > > > > > callcounter=yes
> > > > > > 
> > > > > > 
> > > > > > # context for SIP/modesto
> > > > > > context modesto {
> > > > > > includes {
> > > > > > vivo;
> > > > > > tim;
> > > > > > oi;
> > > > > >   

Re: [asterisk-users] Problem with Atxfer for the calling party

[Roberto Linck]

Hi Antonio,

I'd never had used extensions.ael but in extensions.conf, using Macro I
always set '__TRANSFER_CONTEXT' to the same context of exten and it works
well.

2011/12/13 Antonio Modesto 

> **
> Hello everybody,
>
> I found that if i write my macro in the extensions.conf (not in ael),
> the atxfer works well, the problem is that ael uses gosub instead of the
> Macro() application, which doesn't change the current context. Does anybody
> know if i can do anything to solve this? I know if i rewrite all my macros
> in the common way, it will work, but that's a lot of coding for me.
>
>
>
> On Mon, 2011-12-12 at 08:57 -0200, Antonio Modesto wrote:
>
> Nothing?
>
>
> On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote:
>
>
>
>
>   Hi There,
>
> I'm still having this problem, Does somebody  know what can be
> happening?
>
>
> Regards.
>
> On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote:
>
> Hello,
>
> The exten is the parameter passed to the macro, which contains the sip
> device name. I'll change the name to another less confusing.
>
> * Alexandre, também sou brasileiro hehe, notei que você já escreveu um
> livro sobre asterisk, será que você poderia me ajudar com esse problema? Já
> tem alguns dias que estou na luta aqui hehe.
>
> On Fri, 2011-11-11 at 08:45 -0200, Alexandre Keller wrote:
>
> You're using ${exten} inside your macro, you should use ${EXTEN}.
> --
> Atenciosamente,
>
> ALEXANDRE KELLER
>
>
> http://twitter.com/alexandrekeller
> http://www.facebook.com/alexandre.keller.BR
>
> "Dinheiro é a consequência de um trabalho bem feito e não o motivo para se
> fazer um bom trabalho."
>
>
> *P Antes de imprimir pense em seu compromisso com o Meio Ambiente.*
>
> On 11/11/2011, at 08:38, Antonio Modesto wrote:
>
>  On Mon, 2011-11-07 at 09:12 -0600, Danny Nicholas wrote:
>
> It can have to do with either the telephones dial plan or the context in
> the Asterisk dial plan combined with your features.conf settings.
>
>
> I noticed that my problem occurs when i use a macro to dial sip devices,
> my dialplan is like this:
>
> - Each sip device has its own context
> - This context includes the outgoing call contexts that this extension can
> use for making calls and includes a context called "ramais", which has the
> dial plan to call another extensions, it uses a macro to do this.
>
> Here is the configuration for my extension "modesto" :
>
> # sip.conf
> [modesto](default_extension)
> username=modesto
> context=modesto
> callerid="modesto" <106>
> callgroup=4
> pickupgroup=4
>
> # Default extension template
> type=friend
> dtmfmode=auto
> host=dynamic
> disallow=all
> allow=ulaw
> allow=alaw
> deny=0.0.0.0/0.0.0.0
> permit=192.168.1.0/255.255.255.0
> canreinvite=yes
> qualify=no
> callcounter=yes
>
>
> # context for SIP/modesto
> context modesto {
> includes {
> vivo;
> tim;
> oi;
> claro;
> vivoddd;
> timddd;
> oiddd;
> claroddd;
> embratel;
> embratel2;
> };
> includes {
> ramais;
> };
> };
>
> # Although the problem is occurring also for others contexts included,
> i'll show only the "ramais" context, which is used to call local extensions:
>
> context ramais {
> 101 => &dial_sip(suporte1);
> 102 => &dial_sip(suporte2);
> 103 => &dial_sip(suporte3);
> 105 => &dial_sip(suporte05);
> 106 => &dial_sip(modesto);
> 107 => &dial_sip(gustavo);
> 108 => &dial_sip(pauloh);
> 109 => &dial_sip(fernanda);
> 111 => &dial_sip(marcos);
> 112 => &dial_sip(thiago);
> 115 => &dial_sip(helder);
> 116 => &dial_sip(atendimento01);
> 117 => &dial_sip(atendimento03);
> 118 => &dial_sip(atendimento02);
> 119 => &dial_sip(marlon);
> 120 => &dial_sip(suporteemp);
> 122 => &dial_sip(telemais);
> 123 => &dial_sip(casagustavo);
> 127 => &dial_sip(manutencao);
> 128 => &dial_sip(guilherme);
> 129 => &dial_sip(marcelo);
> 130 => &dial_sip(rafael);
> 132 => &dial_sip(netita2);
> 133 => &dial_sip(unotel);
>
> };
>
> If I use the Dial() application instead of this macro, it works well. I
> noticed that when I use the macro and try to transfer a call (The problem
> occurs only for the calling party, the called party can do transfers with
> no problems), asterisk tries to find the extension in the 
> context and of course, there is no dialplan to call the extensions there.
>
>
> Here is the dial_sip macro:
>
> macro dial_sip(exten) {
> Verbose(2,"==> Chamando a MACRO dial_sip - ponto 1 macros.ael
> <==");
> Verbose(4,"> Macro dial_sip iniciada.");
> ChanIsAvail(SIP/${exten});
> Verbose(2,"==> ${AVAILORIGCHAN}");
>
> if ("${AVAILORIGCHAN}" != "")
> 

[asterisk-users] AEL x LUA

[Antonio Modesto]

Hello,

I would like to receive some suggestions about dialplans written in
lua, actually my dialplan is written in ael, but i'm having some
problems with it. I noticed that asterisk just translates ael to the old
extension language, does it do the same with lua?

Thanks
--
_
-- 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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   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problem with Atxfer for the calling party

[Antonio Modesto]

Hello everybody,

I found that if i write my macro in the extensions.conf (not in
ael), the atxfer works well, the problem is that ael uses gosub instead
of the Macro() application, which doesn't change the current context.
Does anybody know if i can do anything to solve this? I know if i
rewrite all my macros in the common way, it will work, but that's a lot
of coding for me.


On Mon, 2011-12-12 at 08:57 -0200, Antonio Modesto wrote:

> Nothing?
> 
> 
> On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote:
> 
> > 
> > 
> > 
> > Hi There,
> > 
> > I'm still having this problem, Does somebody  know what can be
> > happening?
> > 
> > 
> > Regards.
> > 
> > On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote:
> > 
> > > Hello,
> > > 
> > > The exten is the parameter passed to the macro, which contains
> > > the sip device name. I'll change the name to another less
> > > confusing.
> > > 
> > > * Alexandre, também sou brasileiro hehe, notei que você já
> > > escreveu um livro sobre asterisk, será que você poderia me ajudar
> > > com esse problema? Já tem alguns dias que estou na luta aqui hehe.
> > > 
> > > On Fri, 2011-11-11 at 08:45 -0200, Alexandre Keller wrote:
> > > 
> > > > You're using ${exten} inside your macro, you should use
> > > > ${EXTEN}.
> > > > -- 
> > > > Atenciosamente,
> > > > 
> > > > ALEXANDRE KELLER
> > > > 
> > > > 
> > > > http://twitter.com/alexandrekeller
> > > > http://www.facebook.com/alexandre.keller.BR
> > > > 
> > > > "Dinheiro é a consequência de um trabalho bem feito e não o
> > > > motivo para se fazer um bom trabalho."
> > > > 
> > > > 
> > > > P Antes de imprimir pense em seu compromisso com
> > > > o Meio Ambiente.
> > > > 
> > > > On 11/11/2011, at 08:38, Antonio Modesto wrote:
> > > > 
> > > > 
> > > > > On Mon, 2011-11-07 at 09:12 -0600, Danny Nicholas wrote:
> > > > > 
> > > > > > It can have to do with either the telephones dial plan or
> > > > > > the context in the Asterisk dial plan combined with your
> > > > > > features.conf settings.
> > > > > 
> > > > > 
> > > > > I noticed that my problem occurs when i use a macro to dial
> > > > > sip devices, my dialplan is like this:
> > > > > 
> > > > > - Each sip device has its own context
> > > > > - This context includes the outgoing call contexts that this
> > > > > extension can use for making calls and includes a context
> > > > > called "ramais", which has the dial plan to call another
> > > > > extensions, it uses a macro to do this.
> > > > > 
> > > > > Here is the configuration for my extension "modesto" :
> > > > > 
> > > > > # sip.conf
> > > > > [modesto](default_extension)
> > > > > username=modesto
> > > > > context=modesto
> > > > > callerid="modesto" <106>
> > > > > callgroup=4
> > > > > pickupgroup=4
> > > > > 
> > > > > # Default extension template
> > > > > type=friend
> > > > > dtmfmode=auto
> > > > > host=dynamic
> > > > > disallow=all
> > > > > allow=ulaw
> > > > > allow=alaw
> > > > > deny=0.0.0.0/0.0.0.0
> > > > > permit=192.168.1.0/255.255.255.0
> > > > > canreinvite=yes
> > > > > qualify=no
> > > > > callcounter=yes
> > > > > 
> > > > > 
> > > > > # context for SIP/modesto
> > > > > context modesto {
> > > > > includes {
> > > > > vivo;
> > > > > tim;
> > > > > oi;
> > > > > claro;
> > > > > vivoddd;
> > > > > timddd;
> > > > > oiddd;
> > > > > claroddd;
> > > > > embratel;
> > > > > embratel2;
> > > > > };
> > > > > includes {
> > > > > ramais;
> > > > > };
> > > > > };
> > > > > 
> > > > > # Although the problem is occurring also for others contexts
> > > > > included, i'll show only the "ramais" context, which is used
> > > > > to call local extensions:
> > > > > 
> > > > > context ramais {
> > > > > 101 => &dial_sip(suporte1);
> > > > > 102 => &dial_sip(suporte2);
> > > > > 103 => &dial_sip(suporte3);
> > > > > 105 => &dial_sip(suporte05);
> > > > > 106 => &dial_sip(modesto);
> > > > > 107 => &dial_sip(gustavo);
> > > > > 108 => &dial_sip(pauloh);
> > > > > 109 => &dial_sip(fernanda);
> > > > > 111 => &dial_sip(marcos);
> > > > > 112 => &dial_sip(thiago);
> > > > > 115 => &dial_sip(helder);
> > > > > 116 => &dial_sip(atendimento01);
> > > > > 117 => &dial_sip(atendimento03);
> > > > > 118 => &dial_sip(atendimento02);
> > > > > 119 => &dial_sip(marlon);
> > > > > 120 => &dial_sip(suporteemp);
> > > > > 122 => &dial_sip(telemais);
> > > > > 123 => &dial_sip(casagustavo);
> > > > > 127 => &dial_sip(manutencao);
> > > > > 128 => &dial_sip(guilherme);
> > > > > 129 => &dial_sip(marcelo);
> > > > > 130 => &dial_sip(rafael);
> > > > > 132 => &dial_sip(netita2);
> > > > >

[asterisk-users] AGI script that uses google's text to speech engine

[Lefteris Zafiris]

Hello,

version 0.3 of the asterisk-googletts AGI script just got released,
most noticeable changes are:
The script can now be used to easily build IVRs.  
Fixed compatibility with asterisk 1.4 and older.
Fixed compatibility with older perl versions(5.8.8).
Better input handling.

The latest release, documentation and dialplan examples can be found
here: http://zaf.github.com/asterisk-googletts/

A big thank you to all the users that contributed with feedback,
bug reports and suggestions.


Lefteris Zafiris

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Re: [asterisk-users] SIP MESSAGE outside calls - state of the art?

[Olivier]

2011/12/13, Bruce B :
> I think it only works with certain soft phones. I tried Aastra and it
> doesn't work.
Does it ?

A long time ago, I could send off-calls SIP messages with sipsak.
Comparing sipsak and asterisk-10 outputs should help to understand
what currently keeps this feature to work with Aastra phones.


 But EyeBeam soft phone receives messages.
>
> -Bruce
>
> On Mon, Dec 12, 2011 at 6:40 PM, Jay R. Worthington <
> jayrworthing...@gmail.com> wrote:
>
>> Hiya,
>>
>> SIP Messaging is implemented in asterisk-10...
>>
>> >The only documentation I can find talks about a patch and is pretty
>> > old:>http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging
>> > 
>>
>> Like anything on voip-info.org it's horrible outdated. I think there's a
>> documentation for the message-routing in docs
>>
>>
>>
>>
>> Regards
>>
>> Jay
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>

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