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: > 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 > > > _______________________________________________ -- 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
