Some requests after my FOP2 experience

FootmanFootman
in Feedback
Hello !

Thank you for your work. I tested FOP2 and you have done great job ! I have some requests for you :
- It would be great to have FOP2 in French. If you don't already have French translation, I can offer my files to you.
- Attended transfer with Asterisk 1.4 (it seems I must patch Asterisk for this so I will test).
- For extensions, queues, trunks and conferences, I would be great that non-existent ones appear in grey.
- For trunks, I see used lines only for outgoing calls, not incoming. Is it normal ?
- Can you adapt extensions lines number with call-limit in sip.conf ?
- With an outgoing call, I see the number I called when it is ringing but I see my SDA number when the call is answered. Is it normal ?
- Monitor two asterisk with same extensions, trunks, ... (already asked but it seems it takes too much load for you)

Best regards,

Nicolas.

Comments

  • nicolasnicolas
    Hi,
    - It would be great to have FOP2 in French. If you don't already have French translation, I can offer my files to you.

    If you don't mind donating your translation files, please send them to me and I will add them to FOP2 tarballs.
    - Attended transfer with Asterisk 1.4 (it seems I must patch Asterisk for this so I will test).

    Yes, you must patch Asterisk. I have made an asterisk.rpm for Elastix 1.5 (not 1.6). I can provide you with that file (use at your own risk) if you want to test.
    - For extensions, queues, trunks and conferences, I would be great that non-existent ones appear in grey.
    What do you mean by non-existent ? There is no "standard" way to check for existence via AMI. If you mean registered/notregistered, for extensions it works provided you have qualify=yes in their configs.
    - For trunks, I see used lines only for outgoing calls, not incoming. Is it normal ?

    Is it normal if you have different pear names for inbound calls. In many configurations I have seen that incoming calls are not matched as the same pear as outbound calls, (and are many times received as anonymous calls). If you fix your peer definition in Asterisk you will see both inbound and outbound calls.
    - Can you adapt extensions lines number with call-limit in sip.conf ?

    Really hard to implement. Currently the line display is global for all extensions and can be tweaked in the presence.js file. Call-limit is also going to be deprecated, and many people set it to 50 or to 10, it would make a really horrible fop2 display.
    - With an outgoing call, I see the number I called when it is ringing but I see my SDA number when the call is answered. Is it normal ?

    It could be... what is SDA number?
    - Monitor two asterisk with same extensions, trunks, ... (already asked but it seems it takes too much load for you)

    Too much load not only for me but to fop2_daemon. It won't be available in the next version for sure... I am thinking on ways to do it but did not come up to a solution yet.

    Best regards,
  • FootmanFootman
    - It would be great to have FOP2 in French. If you don't already have French translation, I can offer my files to you.

    If you don't mind donating your translation files, please send them to me and I will add them to FOP2 tarballs.

    No problem. The mail is gone.
    - Attended transfer with Asterisk 1.4 (it seems I must patch Asterisk for this so I will test).

    Yes, you must patch Asterisk. I have made an asterisk.rpm for Elastix 1.5 (not 1.6). I can provide you with that file (use at your own risk) if you want to test.

    No, I use last Bristuff for Asterisk 1.4, not an automated gui. Thanks.
    - For extensions, queues, trunks and conferences, I would be great that non-existent ones appear in grey.

    What do you mean by non-existent ? There is no "standard" way to check for existence via AMI. If you mean registered/notregistered, for extensions it works provided you have qualify=yes in their configs.

    I mean a phone not defined in sip.conf, for example. But you said it's not possible in "standard" way. Is it possible that a trunk appears red when it is unreachable or not registered ?
    - For trunks, I see used lines only for outgoing calls, not incoming. Is it normal ?

    Is it normal if you have different pear names for inbound calls. In many configurations I have seen that incoming calls are not matched as the same pear as outbound calls, (and are many times received as anonymous calls). If you fix your peer definition in Asterisk you will see both inbound and outbound calls.

    OK, I'll see that.
    - Can you adapt extensions lines number with call-limit in sip.conf ?

    Really hard to implement. Currently the line display is global for all extensions and can be tweaked in the presence.js file. Call-limit is also going to be deprecated, and many people set it to 50 or to 10, it would make a really horrible fop2 display.

    Yes, I understand this is not a good idea.
    - With an outgoing call, I see the number I called when it is ringing but I see my SDA number when the call is answered. Is it normal ?

    It could be... what is SDA number?

    Sorry, I used a French acronym. SDA means Direct Inward Dialing or Direct Dial-In.
    - Monitor two asterisk with same extensions, trunks, ... (already asked but it seems it takes too much load for you)

    Too much load not only for me but to fop2_daemon. It won't be available in the next version for sure... I am thinking on ways to do it but did not come up to a solution yet.

    No problem.

    Thanks for all. :)

    Best regards,

    Nicolas.
  • FootmanFootman
    Hi,
    - For trunks, I see used lines only for outgoing calls, not incoming. Is it normal ?

    Is it normal if you have different pear names for inbound calls. In many configurations I have seen that incoming calls are not matched as the same pear as outbound calls, (and are many times received as anonymous calls). If you fix your peer definition in Asterisk you will see both inbound and outbound calls.

    This is an extract of my sip.conf :

    register => aaaaa:bbbbb@x.x.x.x
    [prov_1]
    type=friend
    host=x.x.x.x
    secret=bbbbb
    username=aaaaa
    context=sda_incoming
    insecure=very
    disallow=all
    qualify=yes
    allow=alaw
    allow=gsm

    And here this is sip channels when two calls are in progress. The first two lines show an inbound call. The other lines show an outbound call.

    1000809001> sip show channels
    Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
    y.y.y.y 401
    00102/00000 0x8 (alaw) No Tx: ACK
    x.x.x.x aaaaa
    00101/00102 0x8 (alaw) No Rx: ACK
    x.x.x.x <ext_num>
    00103/00000 0x8 (alaw) No Tx: ACK
    z.z.z.z 400
    00101/00002 0x8 (alaw) No Rx: ACK

    In FOP2, I see only "1 lines used" for my trunk prov_1. I don't understand what's the problem ?

    Best regards,

    Nicolas.
  • FootmanFootman
    - Attended transfer with Asterisk 1.4 (it seems I must patch Asterisk for this so I will test).

    Yes, you must patch Asterisk.I have made an asterisk.rpm for Elastix 1.5 (not 1.6). I can provide you with that file (use at your own risk) if you want to test.

    I patched my Asterisk but attended transfer still not work. FOP2 still do blind transfer. The patch seems to be applicated :

    > show manager command Atxfer
    Action: Atxfer
    Synopsis: Attended transfer
    Privilege: call,all
    Description: do attended transfer.
    Variables: (Names marked with * are required)
    *Channel: transferer Channel
    *Exten: Extension to transfer to
    Context: Context to transfer to
    ActionID: Optional Action id for message matching.

    I have enough access to the manager. Do I have a configuration problem or a parameter missing ? Finally, I would have your RPM file for test if you can.

    Thanks.

    Nicolas.
  • nicolasnicolas
    Hi Nicolas

    For testing attendant transfers with 1.4 you have to use fop2 beta and add to fop2.cfg: 
    supervised_transfer = 1

    The beta can be downloaded from:

    http://www.fop2.com/downloads/fop2.tgz (centos 5, 32 bits)

    or

    fop2-beta-centos4-32.tgz
    fop2-beta-centos5-64.tgz
    fop2-beta-etch32.tgz
    fop2-beta-etch64.tgz
    fop2-beta-lenny32.tgz
    fop2-beta-lenny64.tgz

    About the number of channels for a trunk, do a "show channels" instead of a "sip show channels" and look for channel names. FOP2 matches on channel names.

    Best regards,

    PS: Thanks for the translation file!
  • FootmanFootman
    Hello !
    For testing attendant transfers with 1.4 you have to use fop2 beta and add to fop2.cfg: 
    supervised_transfer = 1

    The beta can be downloaded from:

    http://www.fop2.com/downloads/fop2.tgz (centos 5, 32 bits)

    or

    fop2-beta-centos4-32.tgz
    fop2-beta-centos5-64.tgz
    fop2-beta-etch32.tgz
    fop2-beta-etch64.tgz
    fop2-beta-lenny32.tgz
    fop2-beta-lenny64.tgz

    OK, thanks but it seems there is a problem with fop2-beta-lenny32.tgz : the fop2_server binary is in 64 bits.
    # /etc/init.d/fop2 start
Starting Flash Operator Panel 2: fop2start-stop-daemon: Unable to start /usr/local/fop2/fop2_server: Exec format error (Exec format error)
# file /usr/local/fop2/fop2_server 
/usr/local/fop2/fop2_server: ELF 64-bit LSB executable, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.8, stripped

    Do you have 32 bits version ?
    About the number of channels for a trunk, do a "show channels" instead of a "sip show channels" and look for channel names. FOP2 matches on channel names.

    Thanks, I understand now. Name between brackets must be same than username in sip.conf. Also, I can have one button for outbound calls and one for inbound calls. Thank you. :)
    PS: Thanks for the translation file!

    No problem. :) You should add the next line in your lang_fr_FR.js beta file :
    lang['parkingslots']       = 'Places de parking';

    Best regards,

    Nicolas.
  • nicolasnicolas
    Hello,

    I have updated the lenny32 beta package. Please try again and let me know.

    Best regards,
  • FootmanFootman
    I have updated the lenny32 beta package. Please try again and let me know.

    Thanks, it works, now. On the other side, attendant transfers seems to work upside :

    01- I connect myself on FOP2 as 400
    02- 401 calls 400
    03- 400 answers
    04- On FOP2, I click on 304 extension button and on the transfer button
    05- 304 rings
    06- 400 has music on hold
    07- 401 waits for an answer
    08- 304 answers
    09- 401 speaks to 304
    10- 401 hangup
    11- 400 speaks to 304

    From step 6 to 11, I think 400 should replace 401 and vice versa, no ? Or I make bad handling ?

    Best regards,

    Nicolas.
  • nicolasnicolas
    Thanks for your detailed reports. I have updated the beta, please redownload it (you can replace only fop2_server) and let me know if it fixes the supervised transfer.

    Best regards,
  • FootmanFootman
    Hi Nicolás.

    Thank you for your responsiveness. It works well, very good job. :)

    Best regards,

    Nicolas.
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