Can you blind transfer the call from a different User Agent (i.e. Polycom
not bria)? Are you sure the ITSp is sending the calls in via port 5080? Can
you call the auto attendant from the outside and transfer a call? If you
cant call the AA and transfer to an extension they are probably sending the
of sipXecs software
(sipx-users@list.sipfoundry.org)
Subject: [sipx-users] One way audio with attended transfers to outside lines
Example: Outside caller calls in to sipx. Call is answered by Bria 3.4.4 user.
Bria user performs a "Call first" (attended) transfer to sales person's cel
Example: Outside caller calls in to sipx. Call is answered by Bria 3.4.4 user.
Bria user performs a "Call first" (attended) transfer to sales person's cell
phone. This places the outside caller on hold and they hear hold music. The
Bria user speaks to the sales person and then chooses "Transfer
I have to make a correction. The external person could hear the
internal person but the internal person could not hear the external
person. Not sure it mattered if the internal person was the caller or
the callee.
Anyway, I've switched to a different SIP server from the ITSP and
hopefully th
I've been having lots of one-way audio issues recently and initially, I
saw some evidence from one sip trace that it might be caused by the
ITSP. I saw the following error in the trace:
X-Asterisk-HangupCause: Switching equipment congestion
X-Asterisk-HangupCauseCode: 42
That particular call f
Can you try with a softphone (like xlite) to see if that phone properly
handles the offer from the provider?
On Mon, Aug 2, 2010 at 6:14 AM, Irena Dolovčak wrote:
> There is no STUN involved, i set the static ip instead.
>
> I have some new information from my provider. They say, the problem is
There is no STUN involved, i set the static ip instead.
I have some new information from my provider. They say, the problem is with
the interoperability; they are using asterisk and the problem starts when
sipx sends the ringback tone. their server doesn't register the call for
10-15 sec and it do
Mike, thanks for your reply.
I have no SIP helpers enabled.. The first thing I do is to turn them off..
So, I have contacted my provider, and they say that their server doesn't
send anything when media should be sent. But i'm really not sure if that's
their problem.. I think it could be that they
oh yes, i forgot to tell.. the snom (on the inside) cannot hear the other
side talking..
On Fri, Jul 30, 2010 at 8:50 AM, Irena Dolovčak wrote:
> Hi,
>
> I have sucessfully established an outbound call, but when i make an inbound
> call, i got just one way audio.. has anybody an idea?
>
> here is
along.
Mike Burden
Lynk Systems, Inc
e-mail: m...@lynk.com <mailto:m...@lynk.com>
Phone: 616-532-4985
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Friday, December 18, 2009 10:50 PM
To: Burden, Mike
Cc: sipx-users
Subject: Re: [sipx-users] One way audio
On Fri, D
On Fri, Dec 18, 2009 at 11:52 AM, Burden, Mike wrote:
> Twice in the past few weeks we have had calls where the audio suddenly
> went “one way.”
>
>
>
> The first time it happened, it was about 30 minutes into the call. It
> happened again today, about 5 minutes into the call.
>
>
>
> The sipX
m...@lynk.com>
Phone: 616-532-4985
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burden,
Mike
Sent: Friday, December 18, 2009 11:52 AM
To: sipx-users
Subject: [sipx-users] One way audio
Twice in the past few weeks we have had
Twice in the past few weeks we have had calls where the audio suddenly
went "one way."
The first time it happened, it was about 30 minutes into the call. It
happened again today, about 5 minutes into the call.
The sipXtrace looks normal:
Time: 2009-12-18T15:51:44.184000Z sipXbridge ->
On Mon, Dec 7, 2009 at 2:09 PM, r...@no-no-badpuppy.com
wrote:
> I have sipXecs 4.0.4 running on a small deployment, all with Polycom phones
> (650 & 670). External calls are routed via a SIP ITSP (vitelity.net).
> Recently, I've noticed that external calls lasting 30 minutes or more will
> b
The one way audio issues discussed recently were when an asterisk gateway
with a PRI card (sangoma) were installed in asterisk and using that as a
gateway. Not an ideal solution in my opinion. The problem went away if the
asterisk solution was backrevved from 1.6 to 1.4.
"sipxbridge uses re-INVITE
I have sipXecs 4.0.4 running on a small deployment, all with Polycom phones
(650 & 670). External calls are routed via a SIP ITSP (vitelity.net).
Recently, I've noticed that external calls lasting 30 minutes or more will
become one way audio right at the 30 minute mark. I don't recall this
h
Hello,
This is an update for the list. Thanks to the support from Ranga this
issue has been tracked down and resolved. The fix is in place and
should be available in the next day or two as part of an update.
Scott
M. Ranganathan wrote:
> On Fri, May 29, 2009 at 1:02 PM, Scott wrote:
>
>
On Fri, May 29, 2009 at 1:02 PM, Scott wrote:
> Hello,
>
> I have been running into some issues lately using various versions of 4.x
> installed with the CentOS ISO. Incoming calls via my ITSP (SIP trunk) work
> great, both ends can hear each other and the quality is great. However,
> outgoing c
Hello,
I have been running into some issues lately using various versions of
4.x installed with the CentOS ISO. Incoming calls via my ITSP (SIP
trunk) work great, both ends can hear each other and the quality is
great. However, outgoing calls have one-way audio issues. I can hear
them fine,
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