[asterisk-users] Set Call type in dial plan
virendra bhati
virbhati at gmail.com
Tue Jan 3 00:29:50 CST 2012
Hi
Might be it will help. Read it and set in extension as per your need.
core show function CHANNEL
-= Info about function 'CHANNEL' =-
[Synopsis]
Gets/sets various pieces of information about the channel.
[Description]
Gets/sets various pieces of information about the channel, additional <item>
may be available from the channel driver; see its documentation for details.
Any <item> requested that is not available on the current channel will
return
an empty string.
[Syntax]
CHANNEL(item)
[Arguments]
item
Standard items (provided by all channel technologies) are:
audioreadformat - R/O format currently being read.
* audionativeformat - R/O format used natively for audio.*
audiowriteformat - R/O format currently being written.
callgroup - R/W call groups for call pickup.
channeltype - R/O technology used for channel.
language - R/W language for sounds played.
musicclass - R/W class (from musiconhold.conf) for hold music.
parkinglot - R/W parkinglot for parking.
rxgain - R/W set rxgain level on channel drivers that support it.
state - R/O state for channel
tonezone - R/W zone for indications played
transfercapability - R/W ISDN Transfer Capability, one of:
SPEECH
DIGITAL
RESTRICTED_DIGITAL
3K1AUDIO
DIGITAL_W_TONES
VIDEO
txgain - R/W set txgain level on channel drivers that support it.
* videonativeformat - R/O format used natively for video*
trace - R/W whether or not context tracing is enabled, only available
*if CHANNEL_TRACE is defined*.
*chan_sip* provides the following additional options:
peerip - R/O Get the IP address of the peer.
recvip - R/O Get the source IP address of the peer.
from - R/O Get the URI from the From: header.
uri - R/O Get the URI from the Contact: header.
useragent - R/O Get the useragent.
peername - R/O Get the name of the peer.
t38passthrough - R/O '1' if T38 is offered or enabled in this channel,
otherwise '0'
rtpqos - R/O Get QOS information about the RTP stream
This option takes two additional arguments:
Argument 1:
'audio' Get data about the audio stream
'video' Get data about the video stream
'text' Get data about the text stream
Argument 2:
'local_ssrc' Local SSRC (stream ID)
'local_lostpackets' Local lost packets
'local_jitter' Local calculated jitter
'local_maxjitter' Local calculated jitter (maximum)
'local_minjitter' Local calculated jitter (minimum)
'local_normdevjitter'Local calculated jitter (normal
deviation)
'local_stdevjitter' Local calculated jitter (standard
deviation)
'local_count' Number of received packets
'remote_ssrc' Remote SSRC (stream ID)
'remote_lostpackets'Remote lost packets
'remote_jitter' Remote reported jitter
'remote_maxjitter' Remote calculated jitter (maximum)
'remote_minjitter' Remote calculated jitter (minimum)
'remote_normdevjitter'Remote calculated jitter (normal
deviation)
'remote_stdevjitter'Remote calculated jitter (standard
deviation)
'remote_count' Number of transmitted packets
'rtt' Round trip time
'maxrtt' Round trip time (maximum)
'minrtt' Round trip time (minimum)
'normdevrtt' Round trip time (normal deviation)
'stdevrtt' Round trip time (standard deviation)
'all' All statistics (in a form suited to
logging, but not for parsing)
rtpdest - R/O Get remote RTP destination information.
This option takes one additional argument:
Argument 1:
'audio' Get audio destination
'video' Get video destination
'text' Get text destination
*chan_iax2* provides the following additional options:
peerip - R/O Get the peer's ip address.
peername - R/O Get the peer's username.
[See Also]
Not available
On Tue, Jan 3, 2012 at 11:53 AM, Faraj Khasib <fkhasib at iconnecths.com>wrote:
> Here is the thing, my sip client can call the same. Extension once as
> audio and once as video, so I cannt turn off video supportat reciever, what
> I guess can be done is in extension.conf , there must be flag or something
> I can manipulate ...
> Sent from my iPhone
>
> On ٠٣/٠١/٢٠١٢, at ٨:١٩ ص, "virendra bhati" <virbhati at gmail.com> wrote:
>
> Which is means like if you are using sip 1234 then give the details of
> [1234] into that open thread and relevent extensions details too
>
> On Tue, Jan 3, 2012 at 11:30 AM, Faraj Khasib <fkhasib at iconnecths.com>wrote:
>
>> Which is?! What I am missing how to set dail plan in extension.conf to
>> pass call type as its .... Not convert request to video
>>
>> Sent from my iPhone
>>
>> On ٠٣/٠١/٢٠١٢, at ٧:٢٩ ص, "virendra bhati" <virbhati at gmail.com> wrote:
>>
>> Hi,
>>
>> Please give you sip phone name and sip.conf and extensions.conf details
>> which is using for that communication.
>> And CLI output of asterisk is also required.
>>
>>
>> On Tue, Jan 3, 2012 at 9:59 AM, Faraj Khasib <fkhasib at iconnecths.com>wrote:
>>
>>> I use asterisk 1.6, my clients are sip clients, I dail using audio call
>>> in my clients but the request is recieved at the other client as video call
>>> request since I am enabling video support for sip
>>>
>>> Sent from my iPhone
>>>
>>> On ٠٢/٠١/٢٠١٢, at ١١:٤٩ م, "Doug Lytle" <support at drdos.info> wrote:
>>>
>>> >
>>> > Faraj Khasib wrote:
>>> >> Please help, I have tried many things I cannt make it work, when I
>>> make an audio call it is converted by asterisk to video call request
>>> >
>>> > Not that I can help, since I don't do any video calling.
>>> >
>>> > But, if you don't give any information about your system (OS and
>>> > version, Asterisk version and what type of phone you are using), you're
>>> > not likely to get much of a response.
>>> >
>>> > Doug
>>> >
>>> >
>>> > --
>>> > Ben Franklin quote:
>>> >
>>> > "Those who would give up Essential Liberty to purchase a little
>>> Temporary Safety, deserve neither Liberty nor Safety."
>>> >
>>> >
>>> > --
>>> > _____________________________________________________________________
>>> > -- 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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>>> --
>>> _____________________________________________________________________
>>> -- 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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>>>
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>>> To UNSUBSCRIBE or update options visit:
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>>
>>
>>
>> --
>>
>> Thanks and regards
>>
>> Virendra Bhati
>> +91-8885268942
>> Software Engineer
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
>
>
>
> --
>
> Thanks and regards
>
> Virendra Bhati
> +91-8885268942
> Software Engineer
>
> --
> _____________________________________________________________________
> -- 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
>
>
> --
> _____________________________________________________________________
> -- 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
>
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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