yes, I think the issue is with the Termination side (provider) but I just 
wanted to make sure before I open a ticket with them..  I have notice I dont 
have that issue with another provider...  I will open up a ticket with them and 
see how it goes...
   
  Thanks
  

Mike Ashton <[EMAIL PROTECTED]> wrote:
  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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