asterisk users - Mar 2005 - Livevoip still no DTMF?

I have a LiveVOIP toll-free DID and DTMF works fine. I haven't had any
complaints so far. Though, I do get a "dead air" (call doesn't
reach my
asterisk box) when I dial the number on rare occasions. A simple re-dial
and it usually works.

Ring back also doesn't work, but I got around that by using MOH to play
a ringing sound, so its not a big deal for me.

I don't specify any DTMF mode in my iax.conf either.

On Thu, 2005-03-31 at 10:33 -0800, Brian Litzinger
wrote:
> I read in the archives a number of discussions about livevoip, DID,
> and DTMF not working.
> 
> However, no resolutions.
> 
> I just setup a livevoip DID and indeed the DTMF does not work.
> 
> The same asterisk context works via broadvoice and via
> direct dialing in to the asterisk server via SIP.
> 
> Just no DTMF with calls via livevoip.
> 
> I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
> 
-- 
Mike Benoit <ipso@snappymail.ca>
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Brian Litzinger wrote:
> Just no DTMF with calls via livevoip.
> 
> I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
Try updating to the latest stable version (1.0.7).  We are using a 
number of LiveVoIP inbound toll-free's and our DTMF is working well.

Robert Jackson

I don't have any difficulty with DTMF with LiveVoip incoming or outgoing.

MARK.

Brian Litzinger wrote:
>I read in the archives a number of discussions about livevoip, DID,
>and DTMF not working.
>
>However, no resolutions.
>
>I just setup a livevoip DID and indeed the DTMF does not work.
>
>The same asterisk context works via broadvoice and via
>direct dialing in to the asterisk server via SIP.
>
>Just no DTMF with calls via livevoip.
>
>I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
>
>  
>

Are you sure you don't mean Ringtone?

W 

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of MF Hulber
Sent: Thursday, March 31, 2005 3:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Livevoip still no DTMF?

I don't have any difficulty with DTMF with LiveVoip incoming or
outgoing.

MARK.

Brian Litzinger wrote:
>I read in the archives a number of discussions about livevoip, DID, and
>DTMF not working.
>
>However, no resolutions.
>
>I just setup a livevoip DID and indeed the DTMF does not work.
>
>The same asterisk context works via broadvoice and via direct dialing 
>in to the asterisk server via SIP.
>
>Just no DTMF with calls via livevoip.
>
>I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
>
>  
>
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> I read in the archives a number of discussions about livevoip, DID,
> and DTMF not working.
> 
> However, no resolutions.
> 
> I just setup a livevoip DID and indeed the DTMF does not work.
> 
> The same asterisk context works via broadvoice and via
> direct dialing in to the asterisk server via SIP.
> 
> Just no DTMF with calls via livevoip.
> 
> I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
Its been working fine here for about a month now. Currently using
CVS-HEAD-03/31/05, however it worked fine with several previous 
cvs-head versions as well.

Below are the pieces I'm using for incoming calls. Might want to
review and compare to whatever you're using. The iax.conf section
is a very basic type=user with a context referring incoming calls
to the liveviop800 section of extensions.conf shown below.

[livevoip800]
include=>bus-ivr-main
exten=>8001234567,1,Dial(${PHONE6}&${PHONE7},10)
exten=>8001234567,2,Goto(bus-ivr-main|s|1)

[bus-ivr-main]
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,20
exten => s,5,Background(npi-greeting)  ; "Thanks for calling press 1
for"

The above essentially rings two Cisco 7960's and if no answer, 
routes the incoming call to bus-ivr-main. The caller can then enter
valid dtmf digits, including allowed four-digit extensions, etc.
Have had zero problems with dtmf.

(Note: the above approach does have an issue with handling ringback to
the caller _after_ they've entered a four-digit extension. That issue
has been documented/discussed on the list, and is associated with
livevoip not handling the iax "ringing" function after a call as been
"s,2,Aanswer". Work arounds have been noted on the list, however
I've
elected not to address it as its just not that big of a deal for us.)

> > I read in the archives a number of discussions about livevoip, DID,
> > and DTMF not working.
> >
> > However, no resolutions.
> >
> > I just setup a livevoip DID and indeed the DTMF does not work.
> >
> > The same asterisk context works via broadvoice and via
> > direct dialing in to the asterisk server via SIP.
> >
> > Just no DTMF with calls via livevoip.
> >
> > I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
>
>Its been working fine here for about a month now. Currently using
>CVS-HEAD-03/31/05, however it worked fine with several previous
>cvs-head versions as well.
>Below are the pieces I'm using for incoming calls. Might want to
>review and compare to whatever you're using. The iax.conf section
>is a very basic type=user with a context referring incoming calls
>to the liveviop800 section of extensions.conf shown below.
>
>[livevoip800]
>include=>bus-ivr-main
>exten=>8001234567,1,Dial(${PHONE6}&${PHONE7},10)
>exten=>8001234567,2,Goto(bus-ivr-main|s|1)
>
>[bus-ivr-main]
>exten => s,1,Wait,1
>exten => s,2,Answer
>exten => s,3,DigitTimeout,5
>exten => s,4,ResponseTimeout,20
>exten => s,5,Background(npi-greeting)  ; "Thanks for calling press 1
for"
This DTMF and livevoip issue seems quite interesting and really mystifies 
me. The fact that some livevoip customers have this issue and others don't, 
makes this all the more confusing.  I love livevoip service and support, I 
think they are great, but this one issue is creating a nightmare for me 
during.

I am currently testing a calling card asterisk application I developed.  I 
have about 30 people presently testing the system for me and the DTMF issue 
has been everyone's main complaint.  It's either the pin number which
they
know they entered correctly is wrong, or the destination number is 
invalid.  Sometimes we get someone on the other line that we did not 
call.   I am really scared of the ramification if I proceed with the 
launching of this service before this issue is resolved.

If Livevoip cannot seem to resolve this problem on their own, is there any 
way that we can put our heads together and get to the bottom of this 
problem?  I am not only referring to those who are experiencing this 
problem, but also to those who have a similar setup and have not 
encountered this problem.  We can compare our settings and any thing else 
we have noticed during testing.  I am going to get the ball rolling.

Here is a summary of what I am doing:

I developed an asterisk calling card application written in C.  This 
application simply prompts for a PIN number, checks to see if the PIN 
number is valid and whether there are sufficient funds to place a call.  If 
there are sufficient funds the application would then prompt for the 
destination number. At least 30%-50% of the time (can be more sometimes), 
one or both of these prompts would be wrong.  Yesterday was a frustrating 
day for me.  After switching over my DIDs from IAX to SIP with the hope 
that this may resolve the problem, I was still consistently able to 
duplicate the problem.  I would immediately switch to a different DID 
provider and get through 100% of the time without any DTMF issues.

Here is a list of things I have setup, what I am doing and what I have 
noticed trying to troubleshoot this problem.

1. I am using a stable version of asterisk CVS-v1-0-03/26/05-16:54:47.

2. I have tested this problem on CVS-HEAD-03/10/05 and CVS-HEAD-03/26/05 
and I am able to duplicate the problem every time.

3. I am using 1800 DID numbers from LiveVOIP.

4. I am using local DID numbers from SixTel

5. I do NOT use any DTMF settings in IAX.conf nor SIP.conf

6. I am using ast_app_getdata to play the requested prompt and store the 
number entered into a variable. I would then display the content of that 
variable in asterisk command line so that I know exactly what values the 
system is receiving and compare it to what was entered.
      Eg. For the PIN Number: reslt = ast_app_getdata(channel, pinprompt, 
pinnum, 10,0);
            For the Destination number: reslt = ast_app_getdata(channel, 
destprompt, destinationno, 16,0);

7. I think (not 100% sure) that the default DigitTimeout is 6 seconds and 
the default ResponseTime is 12 seconds for the ast_app_getdata.


8. When getting data for the PIN number and the destination number I have 
noticed the DTMF issue is more consistent with the destination number 
rather than the PIN number.  My assumption is that the PIN number is always 
10 digits, so there is no pause after the last digit has been 
entered.  Once the system gets 10 digits, it proceeds to the next step 
ignoring any other digits that may have been received by the system.

There is no fixed length for phone numbers; therefore, depending on where 
you are calling, the length of the phone number tends to vary.  After 
entering a 10 digit number the system may receive added data making the 
number invalid.

9.  When I take a look at the content of the variables compared with the 
digits entered, the problem is either one of two things:
A. Double digits in the number, eg. 4076831234 might be 40076831234 or 
40776831234 etc.
B.  Some of the numbers spill over into the next prompt for data.

10. You can call using a local number from Sixtel or a 1800 number from 
LiveVOIP.  We presently do NOT experience any problems with our local DID 
numbers from Sixtel.

11. The problem happens on both IAX and SIP based DID service.

Thanks everyone, that's all I have for now. I really hope we can resolve
this.

> > I read in the archives a number of discussions about livevoip, DID,
> > and DTMF not working.
> > 
> > However, no resolutions.
> > 
> > I just setup a livevoip DID and indeed the DTMF does not work.
> > 
> > I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
> 
> Thanks to all who responded.
> 
> DTMF still doesn't work.   Basically people reported that their
> 800 DID IAX or SIP all worked fine.
> 
> extensions.conf:
> 
> [from-broadvoice]
> exten => s,1,Goto,IVR|s|1
> 
> [from-livevoip]
> exten => 1234567890,1,Goto,IVR|s|1
> 
> [from-sipmedia]
> exten => s,1,Goto,IVR|s|1
> 
> [IVR]
> exten => s,1,Wait,1
> exten => s,2,Answer
> exten => s,3,DigitTimeout,5             ; Set Digit Timeout to 5 seconds
> exten => s,4,ResponseTimeout,10         ; Set Response Timeout to 10
seconds
> exten => s,5,BackGround(welcome)
> 
> 
> Dialing in from broadvoice, sipmedia, and sip directly to the server
> all work fine.  (all are SIP).  The IAX connection from livevoip does
> not work from whatever source you might dial in.
> 
> 
> Perhaps it has something to do with the recent changes that occured
> at livevoip, or perhaps I am just unlucky.
> 
> iax.conf:
> 
> [general]
> bandwidth=high
> allow=all
> jitterbuffer=no
> tos=low
> register => 1234567:01234567890@217.160.244.186
> 
> [livevoip]
> type=friend
> secret=1234567890
> deny=0.0.0.0/0.0.0.0
> permit=217.160.244.186/255.255.255.0
> context=from-livevoip
As noted in my previous post, I'm configured just like you have
shown above with the exception that I also have:
 type=user
 disallow=all
 allow=gsm
 trunk=no
in the above context. Limiting the session to gsm _might_ influence
the form of dtmf that is allowed, but that's a pure guess.

Also, I'm using current CVS-Head (not v1.x) code. If I recall
correctly, seems others have complained about various dtmf
problems somewhere around v1.06/v1.07.

If the allow=gsm doesn't impact your problem, then I'd try
the cvs-head code to see what impact it has.

Brian Litzinger wrote:
  > iax.conf:
> 
> [general]
> bandwidth=high
> allow=all
> jitterbuffer=no
> tos=low
> register => 1234567:01234567890@217.160.244.186
> 
> [livevoip]
> type=friend
> secret=1234567890
> deny=0.0.0.0/0.0.0.0
> permit=217.160.244.186/255.255.255.0
> context=from-livevoip
> 
> sip.conf:
> I have dtmfmode=inband for both sip.media.com and sip.broadvoice.com
> and both are limited to ulaw, alaw.
> 
> 
> 

Get rid of the bandwidth= statement.  In the [livevoip] put disallow=all 
and allow=ulaw (or the ONE codec you want to use).  Also comment out the 
tos=low just to see if that makes any difference.

On April 1, 2005 01:44 pm, Brian Litzinger wrote:
> Made the suggested changes.  Called in via SIP and Cell Phone.  Still
> no response to DTMF.
It's time to get lowlevel.

iax2 debug and look for "received DTMF digit '3'" or
something.  tethereal
will also show you the IAX2 IEs for DTMF. 

If you do not see this, the far side is not sending DTMF, and you need to 
complain to livevoip.  IAX2 DTMF is *always* out of band.

-A.

Brian Litzinger wrote:
> On Fri, Apr 01, 2005 at 12:12:57PM -0600, Eric Wieling aka ManxPower wrote:
> 
>>Brian Litzinger wrote:
>> > iax.conf:
>>
>>>[general]
>>>bandwidth=high
>>>allow=all
>>>jitterbuffer=no
>>>tos=low
>>>register => 1234567:01234567890@217.160.244.186
>>>
>>>[livevoip]
>>>type=friend
>>>secret=1234567890
>>>deny=0.0.0.0/0.0.0.0
>>>permit=217.160.244.186/255.255.255.0
>>>context=from-livevoip
>>>
>>>sip.conf:
>>>I have dtmfmode=inband for both sip.media.com and sip.broadvoice.com
>>>and both are limited to ulaw, alaw.
>>
>>Get rid of the bandwidth= statement.  In the [livevoip] put disallow=all
>>and allow=ulaw (or the ONE codec you want to use).  Also comment out the
>>tos=low just to see if that makes any difference.
> 
> 
> By your command...
> 
> Made the suggested changes.  Called in via SIP and Cell Phone.  Still
> no response to DTMF.
> 
It was worth a try. 8-)  Try allow=gsm instead, but I doubt it will make 
any difference.  Your other option is to just switch providers.

> >> > iax.conf:
> >>
> >>>[general]
> >>>bandwidth=high
> >>>allow=all
> >>>jitterbuffer=no
> >>>tos=low
> >>>register => 1234567:01234567890@217.160.244.186
> >>>
> >>>[livevoip]
> >>>type=friend
> >>>secret=1234567890
> >>>deny=0.0.0.0/0.0.0.0
> >>>permit=217.160.244.186/255.255.255.0
> >>>context=from-livevoip
> >>>
> >>>sip.conf:
> >>>I have dtmfmode=inband for both sip.media.com and
sip.broadvoice.com
> >>>and both are limited to ulaw, alaw.
> >>
> >>Get rid of the bandwidth= statement.  In the [livevoip] put
disallow=all
> >>and allow=ulaw (or the ONE codec you want to use).  Also comment
out the
> >>tos=low just to see if that makes any difference.
> > 
> > 
> > By your command...
> > 
> > Made the suggested changes.  Called in via SIP and Cell Phone.  Still
> > no response to DTMF.
> > 
For those trying to diagnose iax2 dtmf problems associated with LiveVoip,
here is the results from "iax2 debug" after I placed a call to my
account.
The ivr answered the incoming call, and I pressed "3000".

This represents a working CVS-HEAD-03/31/05 with no dtmf problems.

Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass: ACK
   Timestamp: 00122ms  SCall: 00008  DCall: 00027 [208.139.204.228:4569]
Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 008 Type: DTMF    Subclass: 3
   Timestamp: 16083ms  SCall: 00207  DCall: 00005 [217.160.244.186:4569]
Tx-Frame Retry[-01] -- OSeqno: 008 ISeqno: 006 Type: IAX     Subclass: ACK
   Timestamp: 16083ms  SCall: 00005  DCall: 00207 [217.160.244.186:4569]
Rx-Frame Retry[ No] -- OSeqno: 006 ISeqno: 008 Type: DTMF    Subclass: 0
   Timestamp: 17083ms  SCall: 00207  DCall: 00005 [217.160.244.186:4569]
Tx-Frame Retry[-01] -- OSeqno: 008 ISeqno: 007 Type: IAX     Subclass: ACK
   Timestamp: 17083ms  SCall: 00005  DCall: 00207 [217.160.244.186:4569]
Rx-Frame Retry[ No] -- OSeqno: 007 ISeqno: 008 Type: DTMF    Subclass: 0
   Timestamp: 17543ms  SCall: 00207  DCall: 00005 [217.160.244.186:4569]
Tx-Frame Retry[-01] -- OSeqno: 008 ISeqno: 008 Type: IAX     Subclass: ACK
   Timestamp: 17543ms  SCall: 00005  DCall: 00207 [217.160.244.186:4569]
Rx-Frame Retry[ No] -- OSeqno: 008 ISeqno: 008 Type: DTMF    Subclass: 0
   Timestamp: 18003ms  SCall: 00207  DCall: 00005 [217.160.244.186:4569]
Tx-Frame Retry[-01] -- OSeqno: 008 ISeqno: 009 Type: IAX     Subclass: ACK
   Timestamp: 18003ms  SCall: 00005  DCall: 00207 [217.160.244.186:4569]
  == CDR updated on IAX2/livevoip@217.160.244.186:4569-5
    -- Executing Dial("IAX2/livevoip@217.160.244.186:4569-5",
"SIP/3000|15") in
new stack
    -- Called 3000
    -- SIP/3000-ee8f is ringing

As you can see, the "Type: DTMF and Subclass: 3" represents me dialing
the digit "3" followed by 000 in the next frames. No problems
whatsoever
and x3000 rang as expected.