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[asterisk-users] DTMF between Asterisk servers.


 
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davies147 at gmail.com
Guest
PostPosted: Tue Apr 08, 2008 10:02 am    Post subject: [asterisk-users] DTMF between Asterisk servers. Reply with quote
I believe that what you described should "just work" with the caveat
that "dtmf=inband" is rarely the right thing to do over SIP, and is
prone to all sorts of DTMF detection and debounce issues.

I assume you've tried calling a POTS endpoint and listening to see if
you get DTMF passed through?

1) You did not give a great deal of information about what the current
situation was, or what investigations you've already tried, which is
probably why no-one felt they could reply.
2) It may also have been because less than 23 hours had elapsed...

Regards,
Steve

On 08/04/2008, Mark Hamilton <mark.h at cage151.com> wrote:
Quote:

I find it hard to believe no one knows, so is it just plain no helping? J

If someone would like to atleast point me in the right direction that will
deal specifically with what I'm asking, that would be appreciated too.

Much thanks.

From: Mark Hamilton [mailto:mark.h at cage151.com]
Sent: April 7, 2008 11:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: DTMF between Asterisk servers.

Hello,

I'm a little confused on DTMF.

A sip peer is registered on two Asterisk servers. No dtmfmode is set for
them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both
register on each other.



A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call
is transferred to Asterisk 2:

RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at 65.xx.xx.10,,tT,)

Where 12351 accepts the call on Asterisk 2, and in some cases, that call is
transferred out to a PSTN number, or wherever, but not within Asterisk
anymore via provider2, dtmf=rfc2833.

When the call comes in, I'd like it to relay DTMF just dandy. How can I do
so?

There is no NAT between the Asterisk servers or in front of them. However,
Asterisk2 has iptables which allows all UDP traffic to/fro Asterisk1. When
Asterisk2 transfers the call to external endpoints, there might be a LAN,
but relative ports are open on those LANs.

Please help.

Thanks in advance,

Mark.
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mark.h at cage151.com
Guest
PostPosted: Wed Apr 09, 2008 11:11 am    Post subject: [asterisk-users] DTMF between Asterisk servers. Reply with quote
No, I tried calling the inbound DID to see if DTMF passes through. And most
times it does, however, it's not being relayed to the Asterisk server 2, and
then to the direct external phoneline.

I tried changing all dtmfmodes for the sip peer, for the inbound DID
provider, and it didn't work, even tried playing with canreinvite, etc.

Hence why my desperate plea for help.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Davies
Sent: April 8, 2008 11:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF between Asterisk servers.

I believe that what you described should "just work" with the caveat
that "dtmf=inband" is rarely the right thing to do over SIP, and is
prone to all sorts of DTMF detection and debounce issues.

I assume you've tried calling a POTS endpoint and listening to see if
you get DTMF passed through?

1) You did not give a great deal of information about what the current
situation was, or what investigations you've already tried, which is
probably why no-one felt they could reply.
2) It may also have been because less than 23 hours had elapsed...

Regards,
Steve

On 08/04/2008, Mark Hamilton <mark.h at cage151.com> wrote:
Quote:

I find it hard to believe no one knows, so is it just plain no helping? J

If someone would like to atleast point me in the right direction that will
deal specifically with what I'm asking, that would be appreciated too.

Much thanks.

From: Mark Hamilton [mailto:mark.h at cage151.com]
Sent: April 7, 2008 11:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: DTMF between Asterisk servers.

Hello,

I'm a little confused on DTMF.

A sip peer is registered on two Asterisk servers. No dtmfmode is set for
them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both
register on each other.



A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the
call
Quote:
is transferred to Asterisk 2:


RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at 65.xx.xx.10,,t
T,)
Quote:

Where 12351 accepts the call on Asterisk 2, and in some cases, that call
is
Quote:
transferred out to a PSTN number, or wherever, but not within Asterisk
anymore via provider2, dtmf=rfc2833.

When the call comes in, I'd like it to relay DTMF just dandy. How can I do
so?

There is no NAT between the Asterisk servers or in front of them. However,
Asterisk2 has iptables which allows all UDP traffic to/fro Asterisk1.
When
Quote:
Asterisk2 transfers the call to external endpoints, there might be a LAN,
but relative ports are open on those LANs.

Please help.

Thanks in advance,

Mark.

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bwentdg at pipeline.com
Guest
PostPosted: Thu Apr 10, 2008 3:43 pm    Post subject: [asterisk-users] DTMF between Asterisk servers. Reply with quote
Just a thought. A while back there was discussion about the merits of
having a product (in that case an O/S) with contracted vendor support
or relying solely on "list" support.
I note in the post below where one responder states
" It may also have been because less than 23 hours had elapsed...".

Different strokes for different folks.... But "23 Hours" is loooong
time in
the production world with no help on a TELCO problem. Just an observation on
how differently folks see things and what folk need to recognize before
they dump their NORTEL etc and jump into "open-source".

Mark Hamilton wrote:
Quote:
No, I tried calling the inbound DID to see if DTMF passes through. And most
times it does, however, it's not being relayed to the Asterisk server 2, and
then to the direct external phoneline.

I tried changing all dtmfmodes for the sip peer, for the inbound DID
provider, and it didn't work, even tried playing with canreinvite, etc.

Hence why my desperate plea for help.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Davies
Sent: April 8, 2008 11:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF between Asterisk servers.

I believe that what you described should "just work" with the caveat
that "dtmf=inband" is rarely the right thing to do over SIP, and is
prone to all sorts of DTMF detection and debounce issues.

I assume you've tried calling a POTS endpoint and listening to see if
you get DTMF passed through?

1) You did not give a great deal of information about what the current
situation was, or what investigations you've already tried, which is
probably why no-one felt they could reply.
2) It may also have been because less than 23 hours had elapsed...

Regards,
Steve

On 08/04/2008, Mark Hamilton <mark.h at cage151.com> wrote:

Quote:
I find it hard to believe no one knows, so is it just plain no helping? J

If someone would like to atleast point me in the right direction that will
deal specifically with what I'm asking, that would be appreciated too.

Much thanks.

From: Mark Hamilton [mailto:mark.h at cage151.com]
Sent: April 7, 2008 11:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: DTMF between Asterisk servers.

Hello,

I'm a little confused on DTMF.

A sip peer is registered on two Asterisk servers. No dtmfmode is set for
them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both
register on each other.



A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the

call

Quote:
is transferred to Asterisk 2:



RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at 65.xx.xx.10,,t
T,)

Quote:
Where 12351 accepts the call on Asterisk 2, and in some cases, that call

is

Quote:
transferred out to a PSTN number, or wherever, but not within Asterisk
anymore via provider2, dtmf=rfc2833.

When the call comes in, I'd like it to relay DTMF just dandy. How can I do
so?

There is no NAT between the Asterisk servers or in front of them. However,
Asterisk2 has iptables which allows all UDP traffic to/fro Asterisk1.

When

Quote:
Asterisk2 transfers the call to external endpoints, there might be a LAN,
but relative ports are open on those LANs.

Please help.

Thanks in advance,

Mark.


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


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