Wondering if anyone on here can help with a niggling issue: One of our
extensions is unable to make attended transfers at all.
The phone in question is an Elmeg ip290, and works fine for direct
transfers. However, on attempting to make an attended transfer, the first
leg succeeds (the inbound call
I have seen a couple of posts related to this, but no workaround.
Setup;
Asterisk 1.2.13 with Polycom IP501 phones
Caller is sent to the queue with the "t" option
Agent is logged in via AgentCallbackLogin on an extension that is in a
context that includes exclusively agent extensions.
Ag
9 jun 2006 kl. 10.18 skrev Nicola Pascelupo:
Hi everybody, sorry for my english but i'm italian and i don't know it
very well.
I'm trying to do a java-program to traduce and notify asterisk
events to
a Tapi program.
I've a problem with call trasfer.
When i transfer a sip user i would like to
Hi everybody, sorry for my english but i'm italian and i don't know it
very well.
I'm trying to do a java-program to traduce and notify asterisk events to
a Tapi program.
I've a problem with call trasfer.
When i transfer a sip user i would like to put his line on hold but i
can't do it. He listen t
For anyone else who might run into this, I got around the transferring to
voicemail problem by putting a "canreinvite=no" line into the section for each
caller's SIP address in sip.conf. Not ideal, but it works.
I also had to add a "dtmfmode=inband" for my Mediatrix 1204 addresses to be
able to a
I've google for hours trying to find a discussion of a similar problem as the
one I'm having, so forgive me if this has come up before. If it has, please
point me in the right direction!
The problem occurs when a caller (A) is transferred by an intermediary party
(B) to voicemail (Voicemail or Vo
If I understand your problem correctly, you have user a setup with vm
box a, and user b with vm box b, when secretary uses local callFWD
from phone a to phone b, vm of b picks up. And you want that if it was
redirected from phone a vm box of a should answer. I think (I never
tested this) that the R
I have an asterisk box setup and dialplan that is something like this:
(t1/pri)
|
[incoming]
1234,1,Dial(SIP/secretary,30,rt)
1234,2,Voicemail([EMAIL PROTECTED])
Now the "t" in the dial lets the sec transfer with # and if the person
transferred to is unavail it goes to their voicemail -- that w
Good day all
Is it possible to transfer sip calls?And how?
I saw transfer in iax.conf?
Thanks
Altus
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It's strange to reply to my own email. So please see below of new problem
with transfers.
- Original Message -
From: Ariel's M-tech account
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 11:55 AM
Subject: [Asterisk-Users] SIP Transfer problem
I have been following a
I have been following and reading about the SIP
problem of transferring calls with Asterisk. I did not see this
problem as having a fix or having a patch for it. I can not
use the # in our system due to IVR systems we access.
Can someone let me know at what stage this is
at. This is a
Blind and assisted transfer work with Cisco 7960 phones.
Blind transfer works fine with Budgetones.
As long as you register to Asterisk.
Jamie Carl wrote:
Ok, just been thinking about this and thought I would ask before
trying it out again.
What is the state of SIP transfers? By this I mean tra
Ok, just been thinking about this and thought I would ask
before trying it out again.
What is the state of SIP transfers? By this I mean
transfers initiated via SIP messages, not via DTMF and
'#'.
Last time I tried, on X-Lite, clicking the transfer button
dropped the call.
Also, are/will
cvs update -r 1.x channels/chan_sip.c
make install
where 'x' is from 1 to 30
version 1.30 is dated 2003-04-02
if not sure check "rcs2log -v |more"
regards
Martin
On Tue, 1 Apr 2003, Russ Beaupre, P.E. wrote:
> A while ago SIP transfer via the # key on a call to a cell phone via
> iconnect was
A while ago SIP transfer via the # key on a call to a cell phone via
iconnect was working. I updated to the current CVS tonight and now that
functionality is gone. Any ideas as to how to enable it again?
Thanks in advance
-russ
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