I have a Cisco 1760V with FXO ports hooked to POTS lines talking SIP to
asterisk 1.8.15.0.
imagining in extensions.conf:
exten = 1,1,Dial(SIP/121)
exten = 2,1,Dial(SIP/121SIP/122)
When a caller dials extension 2 /and/ I have
trustrpid=yes
generaterpid=yes
sendrpid=yes
in sip.conf and I use
17 aug 2012 kl. 03:15 skrev Phillip Frost:
On Aug 16, 2012, at 6:25 PM, Tiago Geada wrote:
forward to a Local extension that has dialplan requiring the
acknowledgement?
On 16 August 2012 21:12, Phil Frost p...@macprofessionals.com wrote:
I'd like to allow my users to forward their
Hi Patrick,
Thanks for your suggestion, even though I added my hostname in the
/etc/hosts, still the problem persists. Also I tried to install in OpenSuse
12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
hanging at modules while starting Asterisk.
Regards,
Gopal.
I have just installed one of these cards with the intention of using it to
send text messages.
O2 and Vodafone PAYG SIM cards worked fine (couldn't make calls or send texts
before putting on some credit, obviously). Orange and Virgin PAYG SIMs keep
showing Network status: Not Registered. I
On 08/13/2012 04:58 PM, Jerry Geis wrote:
On 08/13/2012 01:13 PM, Jerry Geis wrote:
I am getting a beep beep beep (like a busy or hangup sound) when I
am using my
AGI to start up a conf. (did not happen with Meetme).
The confbridge works, but the beep beep beep is mixed in with the
audio.
On 08/17/2012 06:08 AM, Eric Wieling wrote:
Has anyone experimented with increasing the DAHDI chunk size in improve fax
reliability? If so, did it help, hurt, or not make any difference?
I haven't found issues related to the DAHDI chunk size. The main thing
which used to hurt FAXing with
In article 502e1e95.6030...@pagestation.com,
Jerry Geis ge...@pagestation.com wrote:
Any ideas on this?
Just trying to eliminate any sounds in the confbridge. I use a call to a
local channel,
then call an AGI to bring others in the call, then speak my message.
Before I speak
my message
On 08/17/2012 02:28 AM, Olle E. Johansson wrote:
If a call is forwarded and hit the dialplan again, it's forwarded to the
context set in the channel variable FORWARD_CONTEXT.
So you could set this variable before you hit queue(), then do things
differently in the context specified by this
Carlos,
I am waiting for my Grandstreams to arrive too.
Similar reasons. Great feature set, reasonable price.
My primary interest is security. Grandstream claims their intermediate
and higher-end models support TLS and SRTP. I am really tired of trying
to make Cisco phones to communicate
On Fri, Aug 17, 2012 at 9:08 AM, Vladimir Mikhelson v...@mikhelson.comwrote:
My primary interest is security. Grandstream claims their intermediate
and higher-end models support TLS and SRTP. I am really tired of trying to
make Cisco phones to communicate securely with Asterisk. Cisco has a
On Fri, 2012-08-17 at 09:30 -0700, Carlos Alvarez wrote:
On Fri, Aug 17, 2012 at 9:08 AM, Vladimir Mikhelson
v...@mikhelson.com wrote:
My primary interest is security. Grandstream claims their
intermediate and higher-end models support TLS and SRTP. I am
really
On Fri, 2012-08-17 at 09:30 -0700, Carlos Alvarez wrote:
On Fri, Aug 17, 2012 at 9:08 AM, Vladimir Mikhelson
v...@mikhelson.com wrote:
My primary interest is security. Grandstream claims their
intermediate and higher-end models support TLS and SRTP. I am
really tired of trying to make Cisco
On 08/17/2012 10:09 AM, Phil Frost wrote:
On 08/17/2012 02:28 AM, Olle E. Johansson wrote:
If a call is forwarded and hit the dialplan again, it's forwarded to
the context set in the channel variable FORWARD_CONTEXT.
So you could set this variable before you hit queue(), then do things
Hello Everyone,
We are trying to integrate a hosted soft-switch to an Asterisks server and
the error received on the Softswitch end is decline 603
The change that we made is to add the Softswitch IP in the SIP
configuration file, see below
[from-trunk]
host=66.77.199.205
type=user
nat=yes
I see no indication of how to do this in sip.conf, and when I start
Asterisk, it doesn't wait on port 80.
Greetings,
--
Juan Carlos Castro y Castro
Instant Solutions - Telefonia Gerando Resultado
http://www.instant.com.br
Principais capitais: 4063-6100
Demais regiões: (11)4063-6100
--
Hello
I think you must change
type = peer
insecure=invite,port
qualify=yes ; for monitor the ip
Regards
On Fri, Aug 17, 2012 at 2:11 PM, Selecstine Bucci Anukwu
buchal...@gmail.com wrote:
Hello Everyone,
We are trying to integrate a hosted soft-switch to an Asterisks server and
the error
On 08/17/2012 06:36 AM, Jerry Geis wrote:
On 08/13/2012 04:58 PM, Jerry Geis wrote:
On 08/13/2012 01:13 PM, Jerry Geis wrote:
I am getting a beep beep beep (like a busy or hangup sound) when I
am using my
AGI to start up a conf. (did not happen with Meetme).
The confbridge works, but the
Hello,
I'm trying to monitor a Call Queue with BLF-button to see if there are
calls inside the call queue.
This I have :
extensions.conf
exten = 566,hint,Queue:voipq1
On the CLI I see :
566@908001-blf : Queue:voipq1State:Unavailable Watchers 1
But when a call enters my queue
On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br wrote:
I see no indication of how to do this in sip.conf, and when I start
Asterisk, it doesn't wait on port 80.
Greetings,
--
Juan Carlos Castro y Castro
Instant Solutions - Telefonia Gerando Resultado
- Original Message -
On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br
wrote:
I see no indication of how to do this in sip.conf, and when I start
Asterisk, it doesn't wait on port 80.
Websocket support is being actively worked on. HTTP support should
be
On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp jc...@digium.com wrote:
- Original Message -
On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br
wrote:
I see no indication of how to do this in sip.conf, and when I start
Asterisk, it doesn't wait on port 80.
On Fri, Aug 17, 2012 at 4:07 PM, Andrew Latham lath...@gmail.com wrote:
On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp jc...@digium.com wrote:
- Original Message -
On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br
wrote:
I see no indication of how to do this in
On Fri, Aug 17, 2012 at 5:01 PM, Joshua Colp jc...@digium.com wrote:
- Original Message -
On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br
wrote:
I see no indication of how to do this in sip.conf, and when I start
Asterisk, it doesn't wait on port 80.
On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro jcas...@instant.com.br wrote:
I still get unauthorized from sipml5 with these modifications. I
used port 80 instead of 8088 (no other webserver listening on 80), was
that wrong?
Correction. It's actually Failed to connect to the server. I set the
I'm trying to monitor a Call Queue with BLF-button to see if there
are calls inside the call queue.
Currently asterisk doesn't support hint's on queues, unless done externally.
See my review 'Support a hint on a queue'
https://reviewboard.asterisk.org/r/1619/
which also
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