13 thoughts on - Set Call type in dial plan

  • 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, Please how to set the call type at extensions.conf, I tried setting the codec manually but didnt work also… any help .. any suggest will be great
    Thanx
    ________________________________________
    Sent: Monday, January 02, 2012 3:07 AM

    Hi All,
    How to set C all type (Audio/Video) in dial plan?
    Regards
    Faraj Khasib

  • Faraj Khasib wrote:

    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

  • 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

  • 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.

  • 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

    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.

    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

  • 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

  • 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

    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

    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

    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.

    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

  • 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
    may be available from the channel driver; see its documentation for details.
    Any
    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

  • this is what my SIP Invite message when I make Video call

    INVITE sip:6500@192.168.21.102 SIP/2.0
    Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport
    Contact: ;+g.oma.sip-im;language=”en,fr”;+g.3gpp.icsi-ref=”urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”
    Call-ID: b9453704-d76a-b8ce-3247-c999abff7395
    CSeq: 324677463 INVITE
    Content-Type: application/sdp
    Content-Length: 588
    Max-Forwards: 70
    Route:
    Accept-Contact: *;+g.3gpp.icsi-ref=”urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”
    P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
    Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER
    Privacy: none
    P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
    User-Agent: Medcor
    Supported: 100rel

    v=0
    o=doubango 1983 678901 IN IP4 192.168.21.193
    s=-
    c=IN IP4 192.168.21.193
    t=0 0
    m=audio 36372 RTP/AVP 8 0 9 101
    a=ptime:20
    a=rtpmap:8 PCMA/8000/1
    a=rtpmap:0 PCMU/8000/1
    a=rtpmap:9 G722/8000/1
    a=rtpmap:101 telephone-event/8000/1
    a=fmtp:101 0-15
    m=video 59296 RTP/AVP 125 106 121 103
    a=rtpmap:125 VP8/90000
    a=fmtp:125 CIF=2;QCIF=2;SQCIF=2
    a=rtpmap:106 H264/90000
    a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; max-mbps=11880
    a=rtpmap:121 MP4V-ES/90000
    a=fmtp:121 profile-level-id=3
    a=rtpmap:103 H263-1998/90000
    a=fmtp:103 CIF=2;QCIF=2;SQCIF=2

    when I make Audio call requests I dont have the video part …. but at receiver since two clients can make video call they have Asterisks adds the Video Part in request sent to receiver,I dont want that part added , how I can delete it ?

  • Hi,

    For such call you just need to select the outbound codec before the dial()
    app.

    choose the audio-only codecs and thus no video codec strings will be
    exchanged in that call.

  • thats excatly what I want, can u plz give me the command, I want to choose only ulow
    ________________________________________
    Sent: Tuesday, January 03, 2012 3:26 AM

    Hi,

    For such call you just need to select the outbound codec before the dial() app.

    choose the audio-only codecs and thus no video codec strings will be exchanged in that call.

  • I already tried what u posted …. didnt work ….
    but thanx for the reply 🙂
    ________________________________________
    Sent: Wednesday, January 04, 2012 11:32 PM

    Hi,
    Sorry for late reply. Hope you’ve already found out something about it.

    What version of asterisk you are using, that function for choosing inbound/outbound call leg codecs is for newer versions of asterisk.
    See these pages:
    http://www.voip-info.org/wiki/view/Asterisk+variables
    https://issues.asterisk.org/view.php?id=13243

    Regards,
    Sammy

    thats excatly what I want, can u plz give me the command, I want to choose only ulow
    ________________________________________
    Sent: Tuesday, January 03, 2012 3:26 AM

    Hi,

    For such call you just need to select the outbound codec before the dial() app.

    choose the audio-only codecs and thus no video codec strings will be exchanged in that call.