similar to: Inband announcement of parking slot from app_parkandannounce?

Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from

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'

Hi List, I need to modify the callerID name of the call coming back when a parked call returns to the extension that parked it when it times out. Looking at app_parkandannounce.c /* Now place the call to the extention */ snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars =

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

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

Hello, I have the following scenario: 1. VoIP Gateway G400 connected to PSTN 2. Asterisk server 1 (working as IVR) 3. Asterisk server 2 (working as ACD, with several agents connected) I have incoming calls coming from PSTN through the VoIP Gateway to Asterisk server 1 (IVR). When the IVR ends working with the call, transfers it to the Asterisk server 2 (ACD). In Asterisk server 1

Alright, I am basically cheap, and I have a cellular plan which allows for free incoming calls (Nextel). I was wondering if there was any way to do sort of a dialback trick in the extensions.conf. I call into the system from my cell phone (maybe via DISA), I dial an internal extension, and dial a phone number. Then * sends to my cellphone the number dialed thus giving me a in call on the cell. Or

How does one do translation for calls that come in from other pbx's where the incoming caller ID is an internal extension number on their pbx? Eg. when I get a call from Free-World-Dial the CID shows up as "429102" which is essentially their internal extension number sans any routing prefix. To dial the number back I need to dial the extension with FWD's routing prefix

Hi! Maybe someone could help me out? When a call is routed via a2billing AGI and user does a transfer, the call is dropped. If the trunk is called directly everyhing works. Here's a direct scenario (working fine): [pbx000001] exten => 101,1,Set(__TRANSFER_CONTEXT=pbx000001) exten => 101,n,Dial(SIP/pozitel/37129238254,45,t) exten => 102,1,Dial(SIP/12345,60) so, when user calls ext

Hello all, When I call the Broadvoice number all is good. When I try to call out through DISA on my broadvoice line i get the following: Executing Dial("SIP/147.135.0.129-0815bc60", "SIP/16037862111@proxy.bos.broadvoice.com|30") in new stack -- Called 16037862111@proxy.bos.broadvoice.com -- Got SIP response 480 "Temporarily Not Available" back from

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

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

I've read the Wiki on using asterisk's built-in transfer options (#8 and #6). They work fine but how does one 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

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

Hello, >The ONLY issue I have is that I don't get ringing dialback so >calling out gives a silence until the other party picks up Have you turned on early B3? S,1,Dial,CAPI/12345678:b${EXTEN}|30 (always early B3) >(plus the recent changes to locks in * required a tweak to the >chan_capi source to match). could you post this tweak here? I'm stuck with CVS-STABLE ;-(

Hello, I'd like to implement something similar to an attended transfer, but with a little more control (I'd like to be able to use MixMonitor and StopMixMonitor to control the call recording, set the account code, etc. I'm on Asterisk 1.4.26. All of the ways I have seen to do this are complicated plans using MeetMe and applicationmap features, and playing with those over the

changeset: 7353:29db5bded574 tag: tip user: kaf24@firebug.cl.cam.ac.uk date: Wed Oct 12 11:15:02 2005 +0100 summary: Fix 64-bit compile warnings in firmware. UP Dell Precision 330, Centos 4.1 dom0. Should XEN even be accessing anything from the source it was built from? Also had all the domU''s toast right after this, they where running for almost 2 days.

I want to have a "personal meetme conference", so when on a call I can transfer the other party to my personal conference with "#7". (I can then make other calls, and dump them into the conference using "#7" as well, then join myself by dialing "7"). Using: exten => 7,1,MeetMe(${CALLERIDNUM}|Mpd) this works as long as I originate the call. However,

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

I have a problem transferring calls multiple times using DTMF sequences (#, *2). The scenario is: Transferee calls Transferor 1 Transferor 1 transfers to Transferor 2 Transferor 2 transfers to Transfer Target When Transferor 2 enters '#' or '*2', Asterisk no longer reacts and the call remains with Transferor 2. I have tested this with Asterisk 11.2 and 11.5 and