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[asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP

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david_nedved at yahoo.com
Guest

Posted: Thu Mar 27, 2008 4:50 am Post subject: [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channe

--- "Darrick Hartman (lists)" <dhartman at djhsolutions.com> wrote:

Quote:

Do yourself a favor and upgrade a Asterisk 1.4 which has a proper
implementation of DTMF. It's likely your SIP provider upgraded to
something which does not recognize the DTMF tones from Asterisk 1.2.

I've upgraded to 1.4.18 (along with zaptel 1.4.9.2) and still
experiencing the same problem (not recognizing DTMF on SIP inbound
calls) as well as new problems. The new problems are much more severe
than the previous problems so I'm starting a new thread with a more
descriptive subject. I've changed sip.conf to eliminate warnings for
new syntax:

insecure=port,invite
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info

Everything else is as-was in sip.conf, extensions.conf, iax.conf,
rtp.conf, voicemail.conf, zapata.conf, zaptel.conf (although I looked
through the new samples and didn't see anything glaring I needed to
change). For the config files I had not changed I took the new sample
files.

Now in addition to not recognizing DTMF on SIP still, asterisk is now
frequently dropping calls when I start to enter DTMF. On console I get
lines such as:

-- Executing [xxxxx at incoming-viatalk:1] Goto("SIP/xxxxx-081ea720",
"incoming|s|1") in new stack
-- Goto (incoming,s,1)
-- Executing [s at incoming:1] Answer("SIP/xxxxx-081ea720", "") in new
stack
-- Executing [s at incoming:2] BackGround("SIP/xxxxx-081ea720",
"/home/dnedved/hello") in new stack
-- <SIP/xxxxx-081ea720> Playing '/home/dnedved/hello' (language
'en')
== Auto fallthrough, channel 'SIP/xxxxx-081ea720' status is 'UNKNOWN'

It's also happening on zaptel channels (although not nearly so
frequently), so it's not a SIP only problem:

[Mar 27 10:42:07] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 18
(Ring Begin)...
[Mar 27 10:42:08] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 2
(Ring/Answered)...
[Mar 27 10:42:12] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 18
(Ring Begin)...
-- Executing [s at line_nl:1] Goto("Zap/4-1", "incoming|s|1") in new
stack
-- Goto (incoming,s,1)
-- Executing [s at incoming:1] Answer("Zap/4-1", "") in new stack
-- Executing [s at incoming:2] BackGround("Zap/4-1",
"/home/dnedved/hello") in new stack
-- <Zap/4-1> Playing '/home/dnedved/hello' (language 'en')
== Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN'
-- Hungup 'Zap/4-1'

I don't know much about asterisk debugging since it has worked so
flawlessly so far, but I would guess that the Auto fallthrough with
status UNKNOWN means that the application that was running died, didn't
set any return code, so asterisk dropped the call? I'm running in
console mode with 5 v's of verbose mode, how do I find more information
about why it's dropping these calls?

Thanks and best regards,

David

david_nedved at yahoo.com
____________________________________________________________________________________
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

dhartman at djhsolutio...
Guest

Posted: Thu Mar 27, 2008 7:16 am Post subject: [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channe

David Nedved wrote:

Quote:

--- "Darrick Hartman (lists)" <dhartman at djhsolutions.com> wrote:

Quote:

There were several things that changed...

Quote:

Now in addition to not recognizing DTMF on SIP still, asterisk is now
frequently dropping calls when I start to enter DTMF. On console I get
lines such as:

-- Executing [xxxxx at incoming-viatalk:1] Goto("SIP/xxxxx-081ea720",
"incoming|s|1") in new stack
-- Goto (incoming,s,1)
-- Executing [s at incoming:1] Answer("SIP/xxxxx-081ea720", "") in new
stack
-- Executing [s at incoming:2] BackGround("SIP/xxxxx-081ea720",
"/home/dnedved/hello") in new stack
-- <SIP/xxxxx-081ea720> Playing '/home/dnedved/hello' (language
'en')
== Auto fallthrough, channel 'SIP/xxxxx-081ea720' status is 'UNKNOWN'

Try adding this line in the general section of extensions.conf

autofallthrough=no

The default behavior in 1.2 was no. In 1.4 it changed to yes. That
will be your simplest fix (without seeing your dialplan). Asterisk is
moving on to the next step in the dialplan before you enter your digits.
You need to have it wait for the digits to be entered.

Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
http://www.djhsolutions.com/wiki

stotaro at totarotechn...
Guest

Posted: Thu Mar 27, 2008 7:32 am Post subject: [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channe

On Thu, Mar 27, 2008 at 8:16 AM, Darrick Hartman (lists)
<dhartman at djhsolutions.com> wrote:

Quote:

David Nedved wrote:

Quote:

--- "Darrick Hartman (lists)" <dhartman at djhsolutions.com> wrote:

Quote:

There were several things that changed...

Quote:

If the DTMF issue works better in 1.2.X and you do not need the
additional features of 1.4.X then you made the right choice going back
to 1.2.X.

People on the list (mainly dev) want you to test, find bugs, jump
through hoops, and post to Mantis (where you bug might just be closed,
or a general feeling of "You are wrong". All of this testing is free
of course due to the "Benefit of the Community".

In the real world, it would serve you better to do what works best for
your business. Don't let the "Dev" guys push you around, do what
makes sense to your business.

Thanks,
Steve Totaro

david_nedved at yahoo.com
Guest

Posted: Thu Mar 27, 2008 9:32 am Post subject: [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channe

--- "Darrick Hartman (lists)" <dhartman at djhsolutions.com> wrote:

Quote:

Try adding this line in the general section of extensions.conf

autofallthrough=no

The default behavior in 1.2 was no. In 1.4 it changed to yes. That
will be your simplest fix (without seeing your dialplan). Asterisk
is
moving on to the next step in the dialplan before you enter your
digits.
You need to have it wait for the digits to be entered.

Thanks for that. I did see that note in UPGRADE.txt but didn't realize
the full importance of it changing the logic of the dialplan. I've got
it set back to "no" and will read the new version of ATFOT to figure
out how to restructure my dialplan.

So now it seems 1.4.18 is doing the same as 1.2.27 -- working for the
most part but completely ignoring DTMF on incoming SIP calls.

Best regards,

David

david_nedved at yahoo.com
____________________________________________________________________________________
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

davies147 at gmail.com
Guest

Posted: Thu Mar 27, 2008 11:00 am Post subject: [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channe

On 27/03/2008, David Nedved <david_nedved at yahoo.com> wrote:

Quote:

So now it seems 1.4.18 is doing the same as 1.2.27 -- working for the
most part but completely ignoring DTMF on incoming SIP calls.

Perhaps you now need to delve deeper. Capture a UDP trace between your
VoIP provider and Asterisk, and another of the same call between
Asterisk and a handset. Do this for an ordinary voice call, no IVR
menus etc etc.

1) Can you hear the DTMF being sent by the far end by the way?

2) If you use Wireshark to do a VoIP call analysis of the traces, do
you receive any DTMF signalling in the RTP stream, or in INFO packets
from your VoIP provider?

I'm sure there is more...

Steve

jra at baylink.com
Guest

Posted: Thu Mar 27, 2008 1:38 pm Post subject: [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channe

On Thu, Mar 27, 2008 at 08:32:28AM -0400, Steve Totaro wrote:

Quote:

People on the list (mainly dev) want you to test, find bugs, jump
through hoops, and post to Mantis (where you bug might just be closed,
or a general feeling of "You are wrong". All of this testing is free
of course due to the "Benefit of the Community".

In the real world, it would serve you better to do what works best for
your business. Don't let the "Dev" guys push you around, do what
makes sense to your business.

Ok, Steve... I understand both sides of this issue, and the one you're
handwaving is "how much did you pay for Asterisk?"

Nothing in life is free, and people who prefer to use the no-Cost
Asterisk as a PBX base instead of paying Nortel mumble-thousand for an
Option 11 still ought to be prepared to invest *something* in their
outcome.

Being a participating member of the open source community; feeding bugs
back to the developers and the like; that's how you 'pay your bill'
when the software doesn't cost anything.

Sure, *everyone's* not *required* to do it.

But people inclined to use Asterisk ought to be figuring some of this
into their value equation. If it's too troublesome...well, buy a box
from someone.

No?

Cheers,
-- jr 'I am not now, nor have I ever been.... a dev' a
--
Jay R. Ashworth Baylink jra at baylink.com
Designer The Things I Think RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274

Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Joseph Stalin)

stotaro at totarotechn...
Guest

Posted: Thu Mar 27, 2008 1:58 pm Post subject: [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channe

On Thu, Mar 27, 2008 at 2:38 PM, Jay R. Ashworth <jra at baylink.com> wrote:

Quote:

On Thu, Mar 27, 2008 at 08:32:28AM -0400, Steve Totaro wrote:

Quote:

I am a user and a high level integrator, none of what you mention
applies to me. Maybe in a lab if I had time...

I run multi million dollar call centers and very demanding PBXs, it is
not in customer's best interest to run buggy code, therefore it is
also not in my best interest.

It is a similar relationship to corporations and their stockholders,
the corp must do what is in the best interest of the shareholder. I
like to call it good business, none of this rebooting daily, weekly,
monthly crap.

Maybe if you lost $26k/hr due to outages, you might feel differently....

Asterisk is a loss leader for the hardware (cards, appliances,
support, ABE) that is why it is free. Otherwise Asterisk would be
vaporware.

Anyways, Asterisk has many costs but I guess you never took Econ 101
or above in college.

I have brought Asterisk to the attention of CSC, The US State Dept,
large corporations, and foreign governments, is that some form of
contribution to the community? I think promotion is a full time job
in some outfits.

By the way, I use the best components to build my systems and my
consulting fee is pretty nice, so you are right, nothing is free.

Thanks,
Steve Totaro

jra at baylink.com
Guest

Posted: Thu Mar 27, 2008 2:34 pm Post subject: [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channe

On Thu, Mar 27, 2008 at 02:58:31PM -0400, Steve Totaro wrote:

Quote:

I am a user and a high level integrator, none of what you mention
applies to me. Maybe in a lab if I had time...

If you are a high-level integrator, then it seems to me you make direct
profit off the backs of the developers you decline to support.

Quote:

I run multi million dollar call centers and very demanding PBXs, it is
not in customer's best interest to run buggy code, therefore it is
also not in my best interest.

Rockwell Galaxy's are great stuff.

Quote:

It is a similar relationship to corporations and their stockholders,
the corp must do what is in the best interest of the shareholder. I
like to call it good business, none of this rebooting daily, weekly,
monthly crap.

Maybe if you lost $26k/hr due to outages, you might feel differently....

Yup. And if I had lots of outages and that was an issue, I might run a
Galaxy and pay the price. But in fact, not such a problem.

Quote:

Asterisk is a loss leader for the hardware (cards, appliances,
support, ABE) that is why it is free. Otherwise Asterisk would be
vaporware.

Well, most of our cards are Sangomas, actually.

Quote:

Anyways, Asterisk has many costs but I guess you never took Econ 101
or above in college.

Clearly, *you* failed reading comprehension. Smile

My entire point was that there are many different costs -- and that you
were shirking the most important one I could see.

Quote:

I have brought Asterisk to the attention of CSC, The US State Dept,
large corporations, and foreign governments, is that some form of
contribution to the community? I think promotion is a full time job
in some outfits.

Sure.

Everyone contributes something different. And thanks.

Smile

Quote:

By the way, I use the best components to build my systems and my
consulting fee is pretty nice, so you are right, nothing is free.

See? We're in violent agreement.

Cheers,
-- jra
--
Jay R. Ashworth Baylink jra at baylink.com
Designer The Things I Think RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274

Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Joseph Stalin)

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