Video Support - Can't get it to work

Discussion in 'General' started by JohnyBeGood, Jul 25, 2008.

  1. JohnyBeGood

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  2. zeus

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    This sip.conf is in /etc/asterisk or @ /var/www/html/admin/modules/core/etc ?

    For this to work you must change the sip.conf in the 2nd path.

    Try it.
     
  3. JohnyBeGood

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    Hi,

    Yes, I've used sip.conf in /etc/asterisk
    Do I add it just below [general] in /var/www/html/admin/modules/core/etc ?
     
  4. zeus

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    Yes ;)
     
  5. JohnyBeGood

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    Ok, I've added bellow to the /var/www/html/admin/modules/core/etc/sip.conf file

    It still says "could not start video" ?
     
  6. zeus

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    Ok try this from my sip.conf

    ; do not edit this file, this is an auto-generated file by freepbx
    ; all modifications must be done from the web gui

    [general]
    ;
    ; enable and force the sip jitterbuffer. If these settings are desired
    ; they should be set in the sip_general_custom.conf file as this file
    ; will get overwritten during reloads and upgrades.
    ;
    disallow=all
    videosupport=yes
    maxcallbitrate=384

    allow=h261
    allow=h263
    allow=h263p
    allow=h264
    allow=ulaw
    allow=alaw
    ; jbenable=yes
    ; jbforce=yes
    nat=yes
    ; These will all be included in the [general] context
    ;
    #include sip_general_additional.conf
    #include sip_general_custom.conf
    #include sip_nat.conf
    #include sip_registrations_custom.conf
    #include sip_registrations.conf

    ; These should all be expected to come after the [general] context
    ;
    #include sip_custom.conf
    #include sip_additional.conf
    #include sip_custom_post.conf

    After save the file do an amportal restart and video support should work.
    To see if the video support is on in the asterisk CLI type "sip show settings".
    let me know.
     
  7. JohnyBeGood

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    Still no joy :(

    I copied your settings into my /var/www/html/admin/modules/core/etc/sip.conf file
     
  8. zeus

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    There must be a problem with symlinks. Copy the same settings to /etc/asterisk/sip.conf.
    Then make an amportal restart and check if video in on.
    The logic says that it must work.
     
  9. zeus

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    here is my sip show settings

    SIP Port: 5060
    Bindaddress: 0.0.0.0
    Videosupport: Yes
    AutoCreatePeer: No
    Allow unknown access: Yes
    Allow subscriptions: Yes
    Allow overlap dialing: Yes
    Promsic. redir: No
    SIP domain support: No
    Call to non-local dom.: Yes
    URI user is phone no: No
    Our auth realm asterisk
    Realm. auth: No
    Always auth rejects: No
    Call limit peers only: Yes
    Direct RTP setup: No
    User Agent: Asterisk PBX
    MWI checking interval: 10 secs
    Reg. context: (not set)
    Caller ID: Unknown
    From: Domain: banasios.no-ip.biz
    Record SIP history: Off
    Call Events: Off
    IP ToS SIP: CS3
    IP ToS RTP audio: EF
    IP ToS RTP video: AF41
    T38 fax pt UDPTL: No
    RFC2833 Compensation: No
    SIP realtime: Disabled
    ippbx*CLI>
    Global Signalling Settings:
    ---------------------------
    Codecs: 0xc (ulaw|alaw)
    Codec Order: ulaw:20,alaw:20
    T1 minimum: 100
    Relax DTMF: No
    Compact SIP headers: No
    RTP Keepalive: 0 (Disabled)
    RTP Timeout: 0 (Disabled)
    RTP Hold Timeout: 0 (Disabled)
    MWI NOTIFY mime type: application/simple-message-summary
    DNS SRV lookup: Yes
    Pedantic SIP support: No
    Reg. min duration 60 secs
    Reg. max duration: 3600 secs
    Reg. default duration: 120 secs
    Outbound reg. timeout: 20 secs
    Outbound reg. attempts: 0
    Notify ringing state: Yes
    Notify hold state: Yes
    SIP Transfer mode: open
    Max Call Bitrate: 384 kbps
    Auto-Framing: No

    Default Settings:
    -----------------
    Context: from-sip-external
    Nat: Always
    DTMF: rfc2833
    Qualify: 2000
    Use ClientCode: No
    Progress inband: Never
    Language: el
    MOH Interpret: default
    MOH Suggest:
    Voice Mail Extension: *97
     
  10. JohnyBeGood

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    Finally got it working! ;)

    I copied
    after disallow=all statement in /etc/asterisk/sip.conf
    Thanks for your help!
    Now it shows that video is supported

     
  11. cruzzmz

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    I have edit my sip.conf and put the above lines in it
    then i do at
    then i do
    i only get this at

    Global Signalling Settings:
    ---------------------------
    Codecs: 0xc (ulaw|alaw)
    Codec Order: ulaw:20,alaw:20
    T1 minimum: 100
    Relax DTMF: No
    Compact SIP headers: No
    RTP Keepalive: 0 (Disabled)
    RTP Timeout: 0 (Disabled)

    anyone could help????
     
  12. zeus

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    do an amportal restart. If this don't work read this post from the beginning and make the steps as described.
     
  13. zeus

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    Good for you.
    In the future maybe you may have some problems with freepbx updates because freepbx use /var/www/html/admin/modules/core/etc/ path.

    Regards ;)
     
  14. JohnyBeGood

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    Thanks for the tip!
     
  15. JohnyBeGood

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    Did you made changes in /etc/asterisk/sip.conf ?
    after changing that file I got it to work.
     
  16. cruzzmz

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    here is the copy of my
    etc/asterisk/sip.conf

    BTW how do i do a amportal restat?
     
  17. zeus

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    From linux command line
    #amportal restart
     
  18. cruzzmz

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    in my sip show settings, it still show no support for video call :(

    eventhough i have done amportal restart

    :S Help !!!!
     
  19. cruzzmz

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    still no luck in making this possible .... anyone pls help
     
  20. JohnyBeGood

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    Make a backup of your sip.conf and replace it with mine. Do a reload and amportal restart afterwards.


     

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