search for: transferee

While on conversation with another party, I dial the atxfer key sequence. Asterisk says "Transfer" then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep' (language 'en') And then in the console of asterisk is wrote : -- Executing Hangup("Transfe...

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

Hi, In 1.6.1, it seems Attended Transfer do not behave exactly behave like Blind Transfer when transferer hangs up before callee answers : - in Blind Transfer, caller (ie transferee) is hearing Ringing tone when callee's phone is ringing - in Attended Transfer, caller (ie transferee) is hearing Music On Hold when callee's phone is ringing - in Attended Transfer, if callee don't answer in a given time frame, call comes back to transferer again Bottom line is you st...

...e extension 169 answers and uses #170#: -- SIP/169-000000ff answered SIP/nhi-riverside-sip-000000fe -- <SIP/169-000000ff> Playing 'pbx-transfer.ulaw' (language 'en') [Dec 10 13:38:03] DEBUG[29980]: features.c:1330 builtin_blindtransfer: transferer=SIP/169-000000ff; transferee=SIP/side-sip-000000fe; lastapp=; lastdata=; chan=SIP/169-000000ff; dstchan= [Dec 10 13:38:03] DEBUG[29980]: features.c:1333 builtin_blindtransfer: TRANSFEREE; lastapp=Dial; lastdata=SIP/169,18,rtT, chan=SIP/nhi-riverside-sip-000000fe; dstchan=SIP/169-000000ff [Dec 10 13:38:03] DEBUG[29980]: feat...

does * support call return? i want when the operator transfers a call if the transferee is busy or doesn't answer the call the call return back to operator again... this feature may be called: call return on busy call return on no answer Paradise Dove

Dear List, I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable attended transfer feature. but i just can't do it work. I've already set "atxfer = *" (and many other combinations) and all extensions on extensions.conf have the t and T option. But when I'm going to test, it doesn't work. Is there any other file that i have to configure in order to

Hi all Any of you guys could point me in the right direction? I need to make that a blind transfer to return to the transferrer when the transferee does not answer. Scenario: . Miss Jane Doe, our front desk attendant, picks up an external call to Mr. Smith; . Miss Doe flashes, dial Mr. Smith's extension and then hangup; . Mr Smith's phone rings until timeout; . At this point, how to return the call to the Miss Doe's extension...

I'm trying to use Asterisk app_parkandannouce to build a global parking pool from within a couple of Norstar PBXes. Right now I can blind transfer calls into the parking lot, but the slot announcement relies on calling back the 'transferee' after the call is parked and I can't pass enough callerid data out from within the PBX to be able to route the call back in (ie. no PRI facility, just cT1). With the Norstar a two-step transfer (Feature-70) can be performed where the caller is placed on hold while the 'trasferee'...

...cancle an attended transfer? Example: I have person on phone, I hit #6 to being att-transfer. I enter Sally's extension. I let it ring for a few seconds. Sally never picks up but her voicemail does. How do I hangup her voicemail and resume the previous call? The example on the wiki assumes the transferee picks up the phone. :/ -Matthew -- ------------------------------------------------------------------------ Matthew Boehm, IT Director Cypress Telecommunications mboehm@cytelcom.com 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044...

Hi, we are using asterisk 1.2.13. When callbacklogin agent transfer call using SIP phone's transfer feature, he is always in busy status and cannot answer any more incoming call from queue until the transferee hang up the call. -- Regards! Liangliang

...D). In Asterisk server 1 (IVR) I'm using the function Transfer() <http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer>, which sends a SIP REFER to the VoIP Gateway, in the way that is explained in RFC 5589 http://tools.ietf.org/html/rfc5589#section-6.1 Where: VoIP Gateway G400 is de Transferee Asterisk server 1 (IVR) is the Transferor And Asterisk server 2 (ACD) is the Transfer Target The SIP transaction is completed correctly, with the difference that there is no "INVITE (hold)" from Transferor to Transferee. In the G400 once finalized the transaction there is no audio: I th...

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

Hi List, Another dumb conversion question (I hope). I installed 10.0 and copied my 1.4 configuration files over. With a few tweaks everything works great except for 1 feature that I specifically went to 10.0 for. When I do an attended transfer, I still get the receptionists caller ID on the transferred phone instead of the incoming callerID. My assumption is that there is some

...OSE[4699] file.c: -- <SIP/pozitel-0000006a> Playing 'pbx-transfer.gsm' (language 'en') [2011-03-13 09:52:05] VERBOSE[4699] res_musiconhold.c: -- Stopped music on hold on SIP/0010333-00000069 [2011-03-13 09:52:05] DEBUG[4699] features.c: transferer=SIP/pozitel-0000006a; transferee=SIP/0010333-00000069; lastapp=; lastdata=; chan=SIP/pozitel-0000006a; dstchan= [2011-03-13 09:52:05] DEBUG[4699] features.c: TRANSFEREE; lastapp=Dial; lastdata=SIP/pozitel/37129238254,60,tTL(34980000:61000:30000), chan=SIP/0010333-00000069; dstchan=SIP/pozitel-0000006a [2011-03-13 09:52:05] DEBUG[4...

Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

...mU''s toast right after this, they where running for almost 2 days. (XEN) DOM24: (file=mm.c, line=462) Non-privileged attempt to map I/O space 00000000 (XEN) (file=grant_table.c, line=1016) Bad flags (0) or dom (0). (NB. expected dom 0) (XEN) (file=grant_table.c, line=791) gnttab_transfer: Transferee has no reservation headroom (51072,51200) or provided a bad grant ref (00000000) or is dying (1) (XEN) file=/usr/src/redhat/SOURCES/xen-unstable.hg/xen/include/asm/mm.h, line=203) Error pfn 0: rd=ffbec080, od=ff195080, caf=80000002, taf=f0000002 (XEN) DOM24: (file=mm.c, line=350) Could not get page...

...ng? S ----------------- Changes to res_parking.c: ... /* Use the non-macro context to transfer the call */ if(strlen(transferer->macrocontext)) transferer_real_context=transferer->macrocontext; else transferer_real_context=transferer->context; + pbx_builtin_setvar_helper(transferee,"TRANSFERER",transferer->name); "new" dialplan: exten => 7,1,Wait,1 exten => 7,2,Answer exten => 7,3,GotoIf($["${TRANSFERER}" = ""]?10) exten => 7,4,agi(synth|You are being transferred to the other parties personal conference room) exten =&g...

This sure seems like it would be simple. Probably can't see the forest for the trees. I need to use the "native" MOH feature on my little WRT to save processor load. I normally don't use MOH but am playing with atxfer and would like to have something to play to the remote transferee. But when I comment out the "default" clause in musiconhold.conf, I get an error when I do the atxfer, "local_ast_moh_start: No class: default" I dread having to go through and prepend "SetMusicOnHold(native-random)" to every single extension in my dialplan. Is th...

There was a discussion a while back about how to pass Calller ID, when forwarding, as either the calling number, or the forwarding number. Had something to do with "scams" IIRC, but could not find in browsing the archives. So, is it in the docs? Starting point or full tilt would be appreciated. joe a.

Hi, When a SIP hardphone is transfering a call while ringing (caller and callee don't speak to each other) using phone's Transfer key, it seems BLINDTRANSFER remains empty. Though I can see a 302 MOVED TEMPORARILY message coming in. Is there a work around or something obvious I'm missing (it's the first time I'm playing with Dialplan transfert features. context mylocal {