[asterisk-users] Question on Aastra phones and Astrisk

[Matt]

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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