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,,t T,) 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080407/fbb94be3/attachment.htm
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,,t T,) 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080408/030eb189/attachment.htm
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:> > 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.