Hi Nicolás,

that sounds good, so I wait for the next beta ;)
Is there a mailinglist or the like where you announce new beta/version?

Kind Regards
Dominique Görsch

Hi,

thanks for your fast reply, but it still don't work. I guess, the problem is that the var is set after the redirect command. The redirect is done and the new channel is allready fired up (and my GotoIf to decide whether it is an transfer or not is done) 'till the AMI answers with Success on the Variable Set.

It is possible, to change the order? So that I can set the var before the Redirect-Command?

10.0.1.219:2916      <= <msg data="98|blindxfer|96|f635ec21a101102c541b82c67e4505a3" />

127.0.0.1            -> Action: Redirect
127.0.0.1            -> Channel: SIP/ys2mAmt1-00030f5c
127.0.0.1            -> Exten: 488
127.0.0.1            -> Context: from-internal
127.0.0.1            -> Priority: 1
127.0.0.1            -> Async: True

127.0.0.1            -> Action: Setvar
127.0.0.1            -> Channel: SIP/ys2mAmt1-00030f5c
127.0.0.1            -> Variable: _BLINDTRANSFER
127.0.0.1            -> Value: SIP/4901-00030f5d

127.0.0.1            <- Response: Success
127.0.0.1            <- Message: Redirect successful

127.0.0.1            <- Event: Unlink
127.0.0.1            <- Privilege: call,all
127.0.0.1            <- Channel1: SIP/ys2mAmt1-00030f5c
127.0.0.1            <- Channel2: SIP/4901-00030f5d
127.0.0.1            <- Uniqueid1: 1382428822.208932
127.0.0.1            <- Uniqueid2: 1382428822.208933
127.0.0.1            <- CallerID1: 00216685828000104
127.0.0.1            <- CallerID2: 490

127.0.0.1            <- Event: Hangup
127.0.0.1            <- Privilege: call,all
127.0.0.1            <- Channel: SIP/4901-00030f5d
127.0.0.1            <- Uniqueid: 1382428822.208933
127.0.0.1            <- CallerIDNum: 490
127.0.0.1            <- CallerIDName: Goersch Dominique
127.0.0.1            <- ConnectedLineNum: 00216685828000104
127.0.0.1            <- ConnectedLineName: <unknown>
127.0.0.1            <- Cause: 16
127.0.0.1            <- Cause-txt: Normal Clearing

127.0.0.1            <- Response: Success
127.0.0.1            <- Message: Variable Set

Kind regards
Dominique Görsch

Thanks for your reply, but it don't work.
[code]10.0.1.219:4319 <= <msg data="98|blindxfer|96|201a5f6c61cca57502cdfe73469638f7" />

127.0.0.1 -> Action: Redirect
127.0.0.1 -> Channel: SIP/ys2mAmt1-0003030c
127.0.0.1 -> Exten: 488
127.0.0.1 -> Context: from-internal
127.0.0.1 -> Priority: 1
127.0.0.1 -> Async: True

127.0.0.1 -> Action: Setvar
127.0.0.1 -> Channel: SIP/ys2mAmt1-0003030c
127.0.0.1 -> Variable: BLINDTRANSFER
127.0.0.1 -> Value: SIP/4901-0003030d
[/code]

The variable will be set after the redirect is initiated on the Channel that is redirecting. But I need the var set on the new call, to see that this call is an redirected call.
If I redirect with the phone, asterisk fires up the new channel with this variable set. If I redirect via FOP/AMI that var is not set. Do you have any idea, how to fire up the new channel with this var set?

Kind regards
Dominique Görsch

Sure, my dialplan adds the Header, to put the own phone in handsfree mode. But the header is also on the second leg, without adding it. Maybe an asterisk-bug?

Hi,

I have a complex dialplan which takes care of the ${BLINDTRANSFER} variable. If this is set, the called party will ring up a few seconds and if unanswered the call goes back to the transfering extension. This works well for months, when the phone does the transfer.

But if I transfer a call via FOP2 (blindtransfer), the variable will not be set, so the destination rings for ever (or 'til the caller hung up). Is there a way, to set this variable, when transfering with FOP2?

Asterisk 1.8, FOP2.27 (same with 2.26).

Kind regards
Dominique Görsch

Dialplan example (See prio 4 and 19ff.):

CLI> dialplan show 490@from-internal                                                                       [553/20005]
[ Context 'from-internal' created by 'pbx_config' ]
  '490' =>          hint: SIP/4901&SIP/4902                       [pbx_config]
  '_XXX' =>         hint: SIP/${EXTEN}                            [pbx_config]
[ Included context 'localphones' created by 'pbx_config' ]
  '490' =>          1. NoOp(=== Interner Double-Ruf ===)          [pbx_config]
                    2. NoOp(CLI:${CALLERID(NUM)}   CLD:490 (SIP/4901 & SIP/4902)) [pbx_config]
                    3. NoOp(STATE:SIP/4901/${DEVICE_STATE(SIP/4901)}   SIP/4902/${DEVICE_STATE(SIP/4902)}) [pbx_config]
                    4. GotoIf($["${BLINDTRANSFER}" != ""]?TRANSFER) [pbx_config]
                    5. GotoIf($["${DB_EXISTS(RUL/${EXTEN})}]?RUL) [pbx_config]
                    6. Set(_PICKUPMARK=${EXTEN})                  [pbx_config]
                    7. Macro(Normalize)                           [pbx_config]
                    8. GotoIf($["${CALLERID(NAME):0:3}"="Uml"]?DIAL) [pbx_config]
                    9. GotoIf($["${CALLERID(NAME):0:3}"="Abw"]?DIAL) [pbx_config]
                    10. GotoIf($["${CALLERID(NAME):0:3}"="Ret"]?DIAL) [pbx_config]
                    11. GotoIf($["${CALLERID(NAME):0:6}"="Wieder"]?DIAL) [pbx_config]
                    12. SIPAddHeader("Alert-Info:<http://www.notused.com>;info=alert-internal") [pbx_config]
                    13. AGI(ast_get_phonebook.php)                [pbx_config]
     [DIAL]         14. NoOp()                                    [pbx_config]
                    15. GotoIf($[ $["${DEVICE_STATE(SIP/4901)}" != "NOT_INUSE"] & $["${DEVICE_STATE(SIP/4901)}" != "UNAVAILABLE"] & $["${DEVICE_STATE(SIP/4901)}" != "INVALID"]]?BUSY) [pbx_config]
                    16. GotoIf($[ $["${DEVICE_STATE(SIP/4902)}" != "NOT_INUSE"] & $["${DEVICE_STATE(SIP/4902)}" != "UNAVAILABLE"] & $["${DEVICE_STATE(SIP/4902)}" != "INVALID"]]?BUSY) [pbx_config]
                    17. Dial(SIP/4901&SIP/4902)                   [pbx_config]
                    18. Goto(HANGUP)                              [pbx_config]
     [TRANSFER]     19. NoOp(TRANS FROM: ${CUT(BLINDTRANSFER,,1)}) [pbx_config]
                    20. GotoIf($[ $["${DEVICE_STATE(SIP/4901)}" != "NOT_INUSE"] & $["${DEVICE_STATE(SIP/4901)}" != "UNAVAILABLE"] & $["${DEVICE_STATE(SIP/4901)}" != "INVALID"]]?RECALL) [pbx_config]
                    21. GotoIf($[ $["${DEVICE_STATE(SIP/4902)}" != "NOT_INUSE"] & $["${DEVICE_STATE(SIP/4902)}" != "UNAVAILABLE"] & $["${DEVICE_STATE(SIP/4902)}" != "INVALID"]]?RECALL) [pbx_config]
                    22. Dial(SIP/4901&SIP/4902,25,m)              [pbx_config]
     [RECALL]       23. NoOp(WIEDERANRUF)                         [pbx_config]
                    24. SIPAddHeader(Diversion: "${EXTEN}" <sip:${EXTEN}@10.9.8.1>;reason=unconditional) [pbx_config]
                    25. Set(CALLERID(NAME)=Ret ${EXTEN})          [pbx_config]
                    26. RetryDial(,5,-1,${CUT(BLINDTRANSFER,,1)},,m) [pbx_config]
                    27. Goto(HANGUP)                              [pbx_config]
     [RUL]          28. NoOp(Anruf umgeleitet zu ${DB(RUL/${EXTEN})}) [pbx_config]
                    29. SIPAddHeader(Diversion: <sip:${EXTEN}@10.9.8.1>;reason=unconditional;screen=yes) [pbx_config]
                    30. Set(CALLERID(NAME)=Uml ${EXTEN})          [pbx_config]
                    31. Set(CONNECTEDLINE(num)=${DB(RUL/${EXTEN})}) [pbx_config]
                    32. Set(CONNECTEDLINE(num-pres)=allowed)      [pbx_config]
                    33. Set(CONNECTEDLINE(name)=Uml ${DB(RUL/${EXTEN})}) [pbx_config]
                    34. Set(CONNECTEDLINE(name-pres)=allowed)     [pbx_config]
                    35. Dial(LOCAL/${DB(RUL/${EXTEN})}@from-internal) [pbx_config]
     [BUSY]         36. Busy()                                    [pbx_config]
                    37. Goto(HANGUP)                              [pbx_config]
     [HANGUP]       38. NoOp(=== STATUS: ${DIALSTATUS} ===)       [pbx_config]
                    39. NoOp(=== HANGUP: ${HANGUPCAUSE} ===)      [pbx_config]
                    40. HangUp()                                  [pbx_config]

Hi,

if I originate a call via FOP2, an autoanswer-header will be added before I call my own extension. This header is on the other call leg also present. Is this a bug in asterisk or a configuration issue?

Kind regards
Dominique Görsch

Hi all,

I'm testing FOP2 with our Asterisk-related PBX and I am really impressed.
When I monitor our SIP-Trunks, FOP2 shows the Channel and the target extension. Is it possible to show source and target instead?

Kind regards
Dominique Görsch