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.

- 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.

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.

- 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.

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.

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.

Hi Nicolás.

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

Best regards,

Nicolas.