[asterisk-users] Question on Aastra phones and Astrisk

[Matt]

I've tried other phones and the issue does not happen.   I've tried a
different IAX provider and it DOES happen... but only if the
jitterbuffer is on on the REMOTE side.  I am currently working with
aastra to try to figure out if this is a phone or asterisk problem.

On 11/9/06, shadowym <shadowym at hotmail.com> wrote:
> That clarifies it!
>
> First the stupid questions to eliminate the possibility of anything besides
> the phones,
>
> Have you connected a different make hardphone or softphone and confirmed
> that works?
>
> Have you tried a different IAX/SIP provider?
>
> -----Original Message-----
> From: Matt [mailto:mhoppes at gmail.com]
> Sent: Wednesday, November 08, 2006 6:11 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk
>
> It only happens when you go from IAX/SIP --> asterisk box --> aastra phone.
> Doesn't happen PSTN --> asterisk box --> aastra phone.
>
> The aastra people have said they believe it is a codec negotiation issue...
> but the newest firmware didn't fix it.... send them packet dumps.
>
> On 11/7/06, shadowym <shadowym at hotmail.com> wrote:
> > Running several Aastra 9133i and 480CT phones with v1.4 firmware
> > CentOS 4.4, Asterisk 1.2.13, Zaptel 1.2.10, Freepbx2.1.3.  Using all
> > default settings
> >
> > I have not seen that problem. I am not exactly sure we are creating
> > those exact same conditions but it sounds like standard extension use
> > to multiple incoming calls correct?  That is all we are doing plus
> > some more complicated outgoing stuff.
> >
> > -----Original Message-----
> > From: Matt [mailto:mhoppes at gmail.com]
> > Sent: Tuesday, November 07, 2006 5:31 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk
> >
> > *bump*  Anyone?
> >
> > On 11/6/06, Curt Shaffer <cshaffer at gmail.com> wrote:
> > > I wanted to add what we have both seen on traffic captures.
> > >
> > > You see Caller 1's RTP stream. Call 2 comes in and you see the
> > > creation of its RTP stream. After Call 2 is put on hold the RTP
> > > stream from Caller 1 disappears without a trace never to return and
> > > this is when the one way audio is happening.
> > >
> > > And I also wanted to add that I am running 1.4.0 firmware for this
> phone.
> > >
> > > Thanks again!
> > >
> > >
> > >
> > > -----Original Message-----
> > > From: Curt Shaffer [mailto:cshaffer at gmail.com]
> > > Sent: Monday, November 06, 2006 6:58 PM
> > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > > Subject: RE: [asterisk-users] Question on Aastra phones and Astrisk
> > >
> > > I'm the friend mentioned here.
> > >
> > > I am using the Aastra 480i CT. It is SIP to my PBX and IAX
> > > termination from the PBX to my provider. My issue has a slight twist
> > > to it but the same result. For instance his is always where as mine
> > > is frequent but not
> > always.
> > > After I got to finally see it first hand today, I had to start over
> > > from Caller 1 5 times to get it to happen again.
> > >
> > > Caller 1 calls in and Person A answers. Caller 2 calls in and Person
> > > B answers. Person B puts caller 2 on hold and audio drops on Caller 1.
> > > So Person A can hear caller 1 but caller 1 cannot hear Person A.
> > >
> > > This happens more often when Call 1 is on the handset and Call 2 is
> > > on the portable or vis a vi, but this is not always the case. It
> > > does happen to 1 set only but just less frequent.
> > >
> > > I have tried carrierinvite=yes and no but this does not change the
> issue.
> > > The phones are behind a router but the external IP of the router is
> > > on the same network as the * box.
> > >
> > > Thanks!
> > >
> > > Curt
> > >
> > > -----Original Message-----
> > > From: asterisk-users-bounces at lists.digium.com
> > > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt
> > > Sent: Monday, November 06, 2006 6:35 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: [asterisk-users] Question on Aastra phones and Astrisk
> > >
> > > Hi,
> > >        Some odd behaviour here.  A friend and I were talking tonight,
> > > and it seems we have both seen the same problem.   We are both using
> > > aastra phones (I am using 9113is).    We have a connection to and from
> > > providers via SIP and IAX.    When I place a call on the local hold of
> > > the phone, and then pick them back up I can hear them, but they can
> > > not hear me.    However, if I park the call, and then pick it up
> > > again, the audio is fine.
> > >       Tonight I tried placing a call on hold using a Sipura/Linksys
> > > ATA (that is just hitting 'flash', which basically puts the call on
> > > local hold and starts music).    The problem did not manifest itself.
> > >
> > > Has anyone else had this issue?  Do you have a fix for it?  It is an
> > > astrisk issue or an aastra issue?
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