Morning,
I have this issue all the time with providers. I have about 20 specific
areacodes I dial into and use automated IVR dialing.
It usually comes down to the end point termination. I have to use 3
different providers and as you discover areacode/exchange issues you
have to add routing to handle it. Lately all my providers can not handle
in session DTMF into the 305 ( Miami ) areacode, seems all 3 of my
providers are using the same termination (Level3) into that market.
Process is try the call, check CLI and make sure all is ok at your end.
Issue a trouble ticket with your provider, now this is where you find
out how good your provider is. Voip-Jet is awful, they don't even
respond to the trouble tickets, but I still use them for some routes,
mainly overseas. Others will send the standard are you connecting to us
ok, check your config response ( like dealing with Rogers ), then
eventually they will do upstream testing. If it worked before and
stopped typically the provider changed their routing and did not test it
completely. They'll either reroute that exchange back to how it was or
issue a trouble ticket of their own with the termination provider.
At this point it will depend on how good your provider is, how
persistent you are on getting a resolution. Or finding and testing
different providers into that market that and deciphering if it works,
the issue with this is most of the termination into an exchange is with
just a few large termination companies. Like Miami all 3 I use all use
Level 3 and are not having any success in resolving yet ( has been 3
weeks ). Fortunately I rerouted and trunked to my asterisk box in Ft
Lauderdale and can use it's PSTN lines out.
Are we having fun yet?
Mike
Martin Glazer wrote:
Which provider are you using? - I'm having the same issue with one of my
providers. All my other providers work OK.
They claim it has something to do with Asterisk and NAT, however I'm not
convinced of that.
Martin
Mohamed Omar wrote:
Thanks Remzi,
I have tried all the options for dtmf on the trunk for this provider
but some issue.. the dtmf tone is not read properly... some
destination through the same provider works OK while other are not
working properly... I'm not sure how to enable dtmf options in the
logger.conf so I just tried it without seeing the output on the
console..
thank you..
*/Remzi Semsettin Turer <[EMAIL PROTECTED]>/* wrote:
You should check with your SIP trunk provider and see what they
support on their end. They may not be supporting RFC2833, you can
try inband or info and see if that helps (just change config part
for the trunk, not the clients)
Also, in logger.conf, you can enable dtmf logging by adding
DTMF
option to console and from the console see more details on DTMF
tones.
*From:* Mohamed Omar [mailto:[EMAIL PROTECTED]
*Sent:* Wednesday, March 28, 2007 10:41 PM
*To:* [email protected]
*Subject:* [on-asterisk] DTMF issue
I been having issues when I can an extenal number that has
IVR. The
key I press are not recognised sometime or are miss read. My
Asterisk sever is connected to PSTN through a SIP trunk to a
provider so am not connected to the PSTN network Directly...
I have no issue with DTMF when I acces Voicemail or any IVR prompt
on my asterisk server, the issue is just when I call an external
number... Currently all my SIP clients are set to DTMF=RFC2833 in
sip conf and on the Devices too...
Codec i'm mainly using is ulaw and the Asterisk version am
using is
Asterisk 1.2.13 svn rev 47264...
I'm wondering if someone has come across the same issue or
has a
workaround..
Thanks
mohamed.
www.phoneip.net
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