Good morning, I just upgraded to the latest cvs (3/23/04 @ about 7:00am MST) and bumped into a little problem... Dialing in from the pstn to sip phones (x-lite softphone on winders and a grandstream handytone-286 ata), I hear the sip phone ring a few times, but hear nothing from the pstn side 'til the timeout. Then the sip phone stops ringing and I hear ringback at the pstn side. Could this be a latent configuration problem on my end that didn't manifest until I upgraded, or is it a problem with this morning's cvs? (not urgent, of course -- just put cvs from a few weeks ago back, and all is well...) Thanks Jeremy Jones
On Tue, 2004-03-23 at 09:12, Jeremy Jones wrote:> I just upgraded to the latest cvs (3/23/04 @ about 7:00am MST) and > bumped into a little problem... > > Dialing in from the pstn to sip phones (x-lite softphone on winders and > a grandstream handytone-286 ata), I hear the sip phone ring a few times, > but hear nothing from the pstn side 'til the timeout. Then the sip > phone stops ringing and I hear ringback at the pstn side.I'm having a similar problem with 0.7.2 but ONLY if I dial multiple destinations at the same time. Here is a copy of my extension section that does NOT provide ringback no matter what I do. In this example the caller hears ringing while the Ringing() app is run, but it goes away when the Dial() app is called, even with "r" on the dial application line. I don't know if chan_local is an issue or if it's dialing multiple destinations at the same time. exten => 3300,1,Answer exten => 3300,2,Ringing exten => 3300,3,Wait(1) exten => 3300,4,Playback(/etc/asterisk/sounds/pls-wait-connect-call) exten => 3300,5,Wait(1) exten => 3300,6,Ringing exten => 3300,7,Wait(1) exten => 3300,8,Dial(Local/3400@extensions&Zap/25,,r) exten => 3300,9,Hangup -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."
> I just upgraded to the latest cvs (3/23/04 @ about 7:00am MST) and > bumped into a little problem... > > Dialing in from the pstn to sip phones (x-lite softphone on winders and > a grandstream handytone-286 ata), I hear the sip phone ring a few times, > but hear nothing from the pstn side 'til the timeout. Then the sip > phone stops ringing and I hear ringback at the pstn side. > > Could this be a latent configuration problem on my end that didn't > manifest until I upgraded, or is it a problem with this morning's cvs? > > (not urgent, of course -- just put cvs from a few weeks ago back, and > all is well...)I ran into the same thing with Cisco 7960. Looks like the logic in the sip channel has changed recently. Add a ",r" to the end of your Dial statements in extensions.conf and the issue should go away.