> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
> > Sent: Monday, February 24, 2014 12:24 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-user
Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Call transfer problem.
>
> Hi all,
>
> I have a user who is having trouble transferring calls, using a Grandstream
> GXP2xxx.
>
> Here's the use case that I've seen:
>
> I call the user
nt: Monday, February 24, 2014 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call transfer problem.
Hi all,
I have a user who is having trouble transferring calls, using a Grandstream
GXP2xxx.
Here's the use case that I've seen:
I call the user
Hi all,
I have a user who is having trouble transferring calls, using a
Grandstream GXP2xxx.
Here's the use case that I've seen:
I call the user from phone A and he answers on phone B.
Then, he hits the transfer button on his phone and dials an extension
that is reachable by him, but not by me,
Hello, I am having a problem with getting call transfer to work.
This is what is happening:-
1) External call comes in on SIP from a DDI provider
2) The call is answered by extension 204
3) Then extension 204 presses the Xfer button and the call is
placed on hold
4) E
Dear ALL
I have asterisk with sip and it is integrated with avaya
through mediant
[*]-[mediant 2000]-E1--[Avaya]
Now i want to call transfer feature in asterisk means transfer call from one
phone 2 another phone how could it possible with asterisk
Regrads
Sa
Can anyone help with the following problem please?
1) On a receptionist's phone (Snom 360 latest firmware), a call is answered.
2) While on this call a second call comes to the phone but she does not answer it.
3) The receptionist makes an attended transfer placing the first caller on hold a
I'm using IAX2 with VP-320I hardphones for remote users. Everything seems to
be working fine except for call transfer. Is this an issue with the IAX2
itself or the phone? If I flash the same phone with SIP, the problem
disappears.
Regards,
Shaun Singh, Manager
Travelwave
1655 Dufferin Street, Sui
Hi,
I try to set up planet VIP-050 with asterisk. Everything works fine
instead of the call transfer. When I press # console says something
like this:
>Oct 5 11:11:20 DEBUG[25104]: chan_sip.c: sip_rtp_read: Oooh,
format changed >to 1024
>Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbct
Adam Robins wrote:
The double-star now works great. If I press it while on a call, I go
into transfer mode. The problem is that the # still works as well!
Shouldn't the atzfer specification turn off the #?
Blind transfers are on '#' by default, so you may need to move them to
another sequen
I do not want to use the default key of '#' for call transfer, because
as we all know, it interferes with many IVRs that require # as a
termination character. I modified features.conf and added:
[featuremap]
atxfer => **
The double-star now works great. If I press it while on a call, I go
into
you need the x or X option to your Dial command. "show application
dial" is your friend ...
cheers
Michael
On Mon, 11 Oct 2004 08:37:36 -0500 (CDT), [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> On Fri, 8 Oct 2004, Michael Nolan wrote:
>
> Hi !
>
> I have checked my asterisk. It contains thi
On Fri, 8 Oct 2004, Michael Nolan wrote:
Hi !
I have checked my asterisk. It contains this patch or thBis patch is
already compiled into it. can you plz guide me as to how i can make use
of it ? I have pressed '#' but it doesnot give me any dial tone. Are there
any special changes that need t
For testing purposes, my dial line is:
Dial(${ARG2},20,tT)
When I call from one machine through asterisk to
another, I can press # from either side and hear "Transfer."
However, from the caller side I can continue on and
put people on hold by dialing '700'.
From the callee side, I can
My dial statement is (for testing purposes):
123,1,Dial(H323/192.168.1.55|20|tT)
When a caller dials extension 123 I can connect and talk without difficulty.
Both the caller and the callee can press # to drop back to asterisk.
The caller can dial an extension and transfer the callee.
When the
o wondering whether, presently some one is implementing this
>feature or not, if no body is doing that, we can
>start on that
>
>Surajee
>
>
> - Original Message -
> From: George Lin
> To: [EMAIL PROTECTED]
> Sent: Wednesday, June 04, 2003 3:36 AM
>
: [EMAIL PROTECTED]
Sent: Wednesday, June 04, 2003 3:36
AM
Subject: RE: [Asterisk-Users] Call
Transfer Problem
so,
What should the call initiator do if s/he wants to transfer the call initiated
by himself/herself, by using flash keypad or what else ?
I
can see such
ansferring
the call, a call may be parked and then pickedup by another user.
The optionnal URL will be sent to the called party if the channel
supportsit.
Surajee
- Original Message -
From:
George Lin
To: [EMAIL PROTECTED]
Sent: Monday, June 02, 2003 1:11 PM
hi All,
We are working on Soft-PBX using Asterisk.
This relates to CALL TRANSFERRING aspects of Asterisk.
We were able to do one type of call transfering,
ie, the called person can transfer the original call to another
person.
but we were unable to do the other, that is, call
initiat
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