Try all three different modes until you find one that works for you.
Usually dtmfmode=rfc2833 will work, with dtmfmode=info being used if
dtmfmode=rfc2833 doesn't work, and dtmfmode=inband as a last choice.
Only modify this for the SIP trunk you use for sipX. Leave everything
else alone otherwise you may inadvertently cause things to go haywire
on your production system.
Phinux Zhang wrote:
Hi Josh
Thank you for the info. I read the sipXbridge page, yes, as you guess,
"dead air" problem has been fixed.
And about AA problem, actually, we have PSTN gateway and IAX gateway
configured on Asterisk, if I use PSTN gateway to dial sipXecs AA, I can
dial exntensions, but if I use IAX gateway, I can't. Just for sure, you
mean if I add dtmfmode=inband on IAX configuation page will solve this
problem, is that right? The aster...@home currently is in production
mode, I wouldn't touch it unless I am sure.
Thanks again.
Regards
Phinux
On Mon, Nov 30, 2009 at 3:22 PM, Josh Patten
<[email protected]>
wrote:
It would appear to me that something is not running through sipXbridge
or there is a misconfiguration with one of your gateways
what type of PSTN gateway are you using, or does it run through
Asterisk?
Did you set Asterisk to point to port 5080 on your sipX box after
configuring sipXbridge? sipXbridge runs on port 5080.
http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration
for more info on using sipXbridge. This should fix the "dead air" after
the "Please hold while I transfer your call" message
I am 99% sure that the AA problems are related to DTMF. try setting
your DTMF settings to one of the options here:
http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode
If you are using an old version of Asterisk (1.2 or so) then you cannot
directly 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 wrote:
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, when I dial
from Asterisk to sipXecs, the AutoAttends(will use AA for short) works
fine, but I can't dial any extension from AA, AA keeps saying the
extension is invalid, but I am sure the extension I dialed is valid,
and AA even can't recognize the number specified on AA configuration
page.
2. I can dial extension from AA when using PSTN line, but after I
dialed one extension, AA said "Please wait....", after that just
silence, I can't hear dial tone, but the phone I dialed ding ring, it's
very strange.
I hope I can fix this today, or we have to switch to other solution
like Trixbox. I am waiting for you, I really hope you professional guys
can help to figure it out. Thanks in advance, and if you need any
information, just let me know, I'll do what I can.
Thank you very much.
Regards
Phinux
|
