es and it is exclusively a problem with
> > Asterisk. Asterisk's DTMF support has always been marginal at best,
> though
> > I
> > believe 1.6 fixed a lot of DTMF issues. Try some of the steps listed in
> > the
> > link I sent you in my last post like turnin
e 1.6 fixed a lot of DTMF issues. Try some of the steps listed in the
> link I sent you in my last post like turning off the IAX jitterbuffer.
>
>
>
> Josh Patten
> Assistant Network Administrator
> Brazos County IT Dept.
> (979) 361-4676
>
>
>
> Phinux Zhang wrote:
I am currently in Sydney Australia and trying to implement sipXecs, I think
it's easy to configure a dial plan, you'd better ask some specific detailed
questions, not such general questions, if you don't know dial plan, you need
to find a phone and read.
On Thu, Dec 3, 2009 at 12:45 AM, ch...@velo
Don't know if it can help you, but please take a look at this:
http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration
That says:
"Typically ITSPs do not handle certain types of SIP requests such as
*REFER*which is used in Call Transfer operations. To implement call
transfer,
/Asterisk+DTMF gives some information
> about DTMF over IAX and a couple of things you can do to try to make it
> work.
>
>
> Josh Patten
> Assistant Network Administrator
> Brazos County IT Dept.
> (979) 361-4676
>
>
>
> Phinux Zhang wrote:
>
> Hi
>
>
y 1.2, has
> known DTMF problems that have caused me headache before, and the solution
> was to create a SIP trunk and specify the DTMF mode that way Asterisk knows
> how to deal with them.
>
> Could you clarify this is the case?
>
> Josh Patten
> Assistant Network Administr
icate between the two, there is NO
> other way. What you are seeing is common with using the wrong type of DTMF
> mode (digit repetition) and you can specify what type of DTMF to use if you
> specify a trunk. Try what I said to try and report the results back.
>
> Phinux Zhang wro
Is there anybody experienced similar problems? Why Auto-Attendant collects
twice for the extension user dialed? Thank you all
Regards
Phinux
On Tue, Dec 1, 2009 at 2:29 PM, Phinux Zhang wrote:
> I think we had a misunderstand here, I didn't not try to integrate Asterisk
> with si
dtmfmode=auto
>
> Remember to set your inbound/outbound routing rules to send the desired
> numbers to sipX.
>
> Josh Patten
> Assistant Network Administrator
> Brazos County IT Dept.
> (979) 361-4676
>
>
>
> Phinux Zhang wrote:
>
>> Hi Josh
>&
I think it's a problem of DNS, did you run configuration test in sipXecs
administration portal? Did you set up your phones and gateways to use DNS
SRV record? I has such problems before when I was evaluating phone systems.
On Tue, Dec 1, 2009 at 12:31 AM, Nitin wrote:
> Scott Lawrence nortel.com
tly connect the two systems, you have to use an "intermediary" like
> sipXbridge due to the inadequacies of Asterisk's SIP stack.
>
> If you need more Asterisk configuration information I will try to oblige,
> as I know a bit about Asterisk.
>
> Phinux Zhang wro
Hi Ranganathan
I don't exactly understand what you mean, could you please give me more
details? Thank you.
Regards
Phinux
On Mon, Nov 30, 2009 at 1:38 PM, M. Ranganathan wrote:
> On Sun, Nov 29, 2009 at 7:51 PM, Phinux Zhang
> wrote:
> > Hello All
> >
> > We
t;
>
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Phinux Zhang
> *Sent:* Sunday, November 29, 2009 4:51 PM
> *To:* sipx-users@list.sipfoundry.org
> *Subject:* [sipx-users] SIP Trunk problem on sipXecs 4.0.4
>
>
Hello All
We are working on deployment of sipXecs 4.0.4 in our company, but we have
the following two problems related with sip trunk, could you please help me
to take a look and give me some suggestions? Thanks in advance for any
advices.
1. We used aster...@home as our production phone system,
14 matches
Mail list logo
