similar to: Cisco 7940 hook-flashing blind xfer.

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

I was wanting to automate entirely a blind transfer. We are not yet using a powerdialler, so when we hit an answermachine we have to manually leave a message. In order to make this a little quicker, I want to leave a standard message on the answermachine. attempt #1. Use the blind transfer feature. set blind transfer to be "**". extension 22 is exten 22 => Goto

Hi all I am having problems with atxfer if I do the extact same thing with blind xfer it works fine when I hit press #2 (defined in conf for atxfer) i get "transfer" I dial the number I want and i get the following on the console -- Playing 'pbx-transfer' (language 'en') -- Executing Dial("Local/18005558355@jesnjer-f97a,2", "/18005558355")

Greetings, folks. I'm having a problem with blind transfers. It seems that, despite not having the T flag set, callers are able to use the blind transfer option. Scenario is this: - Asterisk 1.2.14 - Caller calls into our call center on one of our many phone numbers. - Call gets placed into queue. - Operator answers call. - Caller is able to hit our blind xfer key sequence (#0) and dial

Does anyone know if the attended transfer in CVS head works with app_queue (and more importantly, chan_agent ?) This is the only thing stopping me from deploying the attended transfer patches. Cheers, Ben -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mark Benson Sent: 16 February 2005 14:44 To: Asterisk

Hello. After doing much web searching and searching archives of this mailing list, I see that my question has been asked at least 6 separate times but no answers have been attached. In a nutshell, is there a way for a SIP phone to easily hook flash a ZAP analog trunk mid-call? (This is important when trying to make use of features on a PSTN analog line such as call waiting, call forwarding,

My SIP users need to transmit the "#" key as part of data entry. Asterisk intercepts and initates a transfer function. I'm almost positive I've seen this discussed somewhere, but none of my searches are finding it. Anyone have a handy answer? Tim McKee -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef

I'm using FreePBX 2.5.2.2 with Asterisk 1.6.1.4. If I make a call and then decide to blind transfer them using ## my side of the call is not hung up. Instead it sends me to voicemail. If somebody calls me and then I blind transfer them with ## I am hung up on as expected. I called from 8678 to 28688. I then transferred the call to 8532. Asterisk acts like it wants to hang up, but then

I am having problems transferring calls from one sip extension to another - the extensions use various phones hardware/software. From what I can tell I should just be able to press # and then dial an extension to blind xfer a call right? How do I do attended xfer? Either the phones (for this test I have tried xlite and budgetone102) are not sending DTMF correctly or something else is amiss...

Hey all, posted this the other day, but re-read it & realized I didn't give enough info to be useful.. so I thought I'd try again.. I'm using AAH 2.0 (* 1.2.0) and am unable to transfer a call when I initiate the outgoing call. In AMPs general settings, I've tried changing the Dial command using tT but transfers are only available when I'm the recipient of the call, not

Hello, Sorry for a bit of a newbie post but we all had to start somewhere right .. I'm wondering if someone can briefly explain the difference between blind and attended transfers and why they would generate two very different CDR entries.? From my own research, it seems that transfers are both ultimately a SIP REFER and thus seeing two different CDR entries just confuses me further.

Hi I'm having difficulties getting up my TE110P (running as a E1) when I connect it to the PRI. If I start the server with a loopback connector everything seems fine and the led is green but when I connect it to the PRI the flashing starts .... I can't seem to find anything in the log that suggests what could be wrong .....Aug 8 11:33:00 DEBUG[1369]: Updated conferencing on 31,

Hi, I just upgraded by 1.0.x home server to 1.2.2. Overall the upgrade went fine, but a strange problem has cropped up with the CALLERID name value of incoming calls from the X101P card. When an incoming call is presented (via Vonage ATA), the calledid value getting double quotes up from: -- Executing NoOp("Zap/1-1", """WIRELESS CALLE" <1404xxxxxxx>") in

Hello. I am a spanish student, so excuse my English. I have this HW: - 2 X100P PCI with two analog lines plugged in. These lines are two extensions of a panasonic PBX. Zap/1 = X100P <-- analog line --> extension #237 PBX Panasonic Zap/2 = X100P <-- analog line --> extension #245 PBX Panasonic - 1 TDM20B with two analog telephones plugged in. Zap/3 = TDM20B

Ist there any possibility to use the funktion "consultative transfer"? ( have 2 ISDN-pones attached to the hfc-nt card, configured as zap) With the "#"-key it ist possible to park the call or to make a "blind transfer" at the moment. I have activated threewaycalling in the zapata.conf file: ; internal S0 bus (first hfc/s card): context=local signalling =

i'm posting this to the bind-users and rsync mailing lists in order to get this archived for the next person who needs to do this :) comments and suggestions are welcome, but please cc: to me because i'm not subscribed to either list. craig ----- Forwarded message from Craig Sanders <cas@taz.net.au> ----- Date: Fri, 27 Jul 2001 10:30:20 +1000 From: Craig Sanders

https://bugzilla.samba.org/show_bug.cgi?id=4320 Summary: Daemon should send stdout, stderr of {pre,post}-xfer command to client Product: rsync Version: 3.0.0 Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P3 Component: core AssignedTo: wayned@samba.org

Kevin P. Fleming wrote: > This is because hook-flash support is in chan_zap, not in the core of > Asterisk. There is no need to support hook-flash on any other channel > type, because every other channel type supported by Asterisk has its own > (more reliable) methods of signaling. I'm seeing many discussions on how to provision Blind Transfer, Attendant Transfer and Call

Hi, I've got an issue since about a week and I still can't solve it after a thorough research. I've setup a rsync server on my backup host, with shares using post/pre-xfer exec options but the scripts seems not to even be called : they should write a lockfile, and write in a logfile some information about the transfer request / report. I'm using rsync through stunnel and

Hi, I have a SIP phone which does not natively support SIP transfers (REFER etc...). So far all that is possible is to enable blind transfers using the t and T arguments in Dial from the # DTMF key. The phone has an R button on it and this can be setup to either send an RFC2833 hook flash message (value 16) or a SIP INFO message which you can edit the contents of (since there seems to be