asterisk users - Oct 2003 - Outgoing call to IVR not being "answered"

I don't know if this is a problem with my cisco sip IP Phones or
asterisk but I thought I would post here in case someone else has
experienced this issue.

When I make a call from my SIP cisco IP Phone to some remote IVRs I
never get the rest of my soft keys, only the "End Call" soft key, and
also DTMF doesn't work... its like the phone is acting like the remote
end hasn't picked up yet.

Here are a couple examples of IVRs that elicit this behavior:

800 327 2177
800 433 7300

Thanks,
David

Ok after find this post
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg07004.htm
l.  I have figured out my issue and I have refined my question.

Does anyone how to make asterisk tell the calling sip-ua that the remote
party "answered" the phone as soon as it sees rtp coming in even
though
the remote IVR didn't "connect" the call.

I found a couple of remote IVRs that when I call them they don't
"connect" the call until the caller chooses an option via DTMF... but
my
cisco phone won't send DTMF until asterisk tells it the call has been
answered

Its american express (800 327 2177) and american airlines (800 433
7300)... i think they are trying to save money by not connecting the
call.

Thanks,
David

>I don't know if this is a problem with my cisco sip IP Phones or
asterisk >but I thought I would post here in case someone else has
experienced this >issue.
>
>When I make a call from my SIP cisco IP Phone to some remote IVRs I
never >get the rest of my soft keys, only the "End Call" soft key,
and
also DTMF >doesn't work... its like the phone is acting like the remote
end hasn't >picked up yet.
>
>Here are a couple examples of IVRs that elicit this behavior:
>
>800 327 2177
>800 433 7300

Juan,

That is a helpful discovery.  If I too route the calls out of an IAX
provider (voicepulse) rather then my usual Cisco SIP Gateway I can get
DTMF through with the Cisco phone still not presenting the usual
"soft"
key options.  

So that means that the Cisco IP Phone IS sending DTMF to calls that are
not yet "connected" and that either the asterisk box or the Cisco SIP
Gateway are the ones not transmitting.

Thanks,
David
>	I see the same problem, I?m trunking 800 through IAXTEL. When I
dial
>American I only see the end call button only but DTMF works; after
>choosing an option I get the rest of the buttons.
>	So I think at least the DTMF problem can be solved.

Howdy,

Does anyone know if there are any problems running Asterisk when using
later 7960 SIP versions like 04.04 or 05.03?

Thanks!

Paul

 
Paul Mahler
pmahler@signate.com
phone: 650-207-9855
fax: 877-408-0105

On Sat, 2003-10-18 at 17:25, Paul Mahler wrote:
> Howdy,
> 
> Does anyone know if there are any problems running Asterisk when using
> later 7960 SIP versions like 04.04 or 05.03?
	I have 4.4 running without problems.

-- 
Juanjo sin .sig

On 18/10/2003 at 18:22 Juan J. Sierralta P. wrote:
>On Sat, 2003-10-18 at 17:25, Paul Mahler wrote:
>> Howdy,
>> 
>> Does anyone know if there are any problems running Asterisk when using
>> later 7960 SIP versions like 04.04 or 05.03?
>
>	I have 4.4 running without problems.
>
I have  05.03 running with no problems...


Andy

I am using 5.03 image on 7940 and 7960 and it is ok

----- Original Message ----- 
From: "Andy Powell" <andy@beagles-den.demon.co.uk>
To: <asterisk-users@lists.digium.com>
Sent: Sunday, October 19, 2003 1:03 PM
Subject: RE: [Asterisk-Users] RE: Are Cisco 7960 SIP versions 4 & 5 safe to
use?

>
> On 18/10/2003 at 18:22 Juan J. Sierralta P. wrote:
>
> >On Sat, 2003-10-18 at 17:25, Paul Mahler wrote:
> >> Howdy,
> >>
> >> Does anyone know if there are any problems running Asterisk when
using
> >> later 7960 SIP versions like 04.04 or 05.03?
> >
> > I have 4.4 running without problems.
> >
>
> I have  05.03 running with no problems...
>
>
> Andy
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

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