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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] > 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/[EMAIL PROTECTED],,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. _______________________________________________ -- 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
