You've lost me
>From what I understand, here is your setup:
PSTN ITSP <--IAX--> aster...@home <--SIP via sipXbridge-->
sipX
If the above scenario is true, then what I said previously stands;
However, if you plan on cutting over to sipX and taking Asterisk out of
the mix completely there shouldn't be any DTMF issues. Asterisk,
especially 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 Administrator
Brazos County IT Dept.
(979) 361-4676
Phinux Zhang wrote:
Thank you Josh
But what I can't understand is, for example, if we use sipXecs as our
phone system, and one of our customer uses aster...@home with the same
version what I am using now, if the customer can't dial our number, I
can't tell him to set up a sip trunk to point to our server or
something, there should be a way to detect the repetition and correct
it, what do you think?
Regards
Phinux
On Tue, Dec 1, 2009 at 3:19 PM, Josh Patten <[email protected]>
wrote:
As I said before, you HAVE to
have a SIP trunk set up on Asterisk
pointing to sipX and vice-versa to properly communicate 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 wrote:
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 <[email protected]>
wrote:
I
think we had a misunderstand here, I didn't not try to integrate
Asterisk with sipXecs using SIP trunk or something, they are just
separate system with different gateways, I just dialed sipXecs
extension from Asterisk, and Asterisk is using IAX trunk from our
provider, and sipXecs using SIP trunk from the same provider.
I found this in the sipxivr.log, you can see attendant collected
digites=11221177, what I dialed is 1217, and for 111011, what I dialed
is 1101, for 44, what I dialed is 4.
"2009-12-01T03:19:24.874000Z":222:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant
Collected digits=111011"
"2009-12-01T03:19:24.874000Z":223:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant
Extension 111011 is not valid"
"2009-12-01T03:19:49.075000Z":224:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant
Collected digits=11221177"
"2009-12-01T03:19:49.075000Z":225:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant
Extension 11221177 is not valid"
"2009-12-01T03:20:06.934000Z":226:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant
Collected digits=44"
"2009-12-01T03:20:06.935000Z":227:sipXivr:INFO:*:Thread-17:00000000:sipxivr:"Attendant::attendant
Extension 44 is not valid"
Regards
Phinux
On Tue, Dec 1, 2009 at 1:51 AM, Josh
Patten <[email protected]>
wrote:
Do
not set it on the IAX trunk, set it on the SIP trunk to your sipX
installation. It doesn't appear you have one set up, and I'm wondering
how you've got it working without a SIP trunk set up to that server.
Create a new SIP trunk and send all the numbers to sipX out that SIP
trunk in your outbound routing. Your SIP trunk should have the
following settings:
host=IP.OF.SIPX.INST
port=5080
type=friend
insecure=invite,port
context=from-internal (check this one, I'm not sure
what AAH uses, or if you even need it)
disallow=all
allow=ulaw
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 tried these four options (rfc2833, info, inband,
auto)
on our IAX trunk, but didn't work for me. We are using aster...@home
2.7 with Asterisk version 1.2.5.
I am not sure if I was modifying on the right place, please see
screenshot, is it right?
Regards
Phinux
On Mon, Nov 30, 2009 at 5:02 PM, Josh Patten < [email protected] <mailto: [email protected]>> wrote:
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]
<mailto: [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
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