If the call is actually answered instead of forwarded does audio establish?
If so please describe how the call is forwarded... aa, sipx userui, soft key from the handset itself? Which is the itsp? On May 4, 2011 5:57 PM, "Joegen Baclor" <[email protected]> wrote: > Call forwarding via UI and call transfer via AA are very much > different. The first one is a mere parallel fork while the second one > is a REFER. You can send me a wireshark capture offlist and I'll see if > we could tell what went wrong from that. > > > On 05/05/2011 05:46 AM, Mike Haun wrote: >> Thanks Tony. That sure seems to fit the bill alright, but what I >> can't figure out is that in both scenarios, the calls are >> outbound because of the result of the call forwarding on the >> extension. Both are outbound from the same extension, but both have >> very different outcomes. I'm just not getting it? >> >> >> On Wed, May 4, 2011 at 3:05 PM, Tony Graziano >> <[email protected] <mailto:[email protected]>> >> wrote: >> >> the call is not being sent to port 5080 by the new itsp... >> >> On Wed, May 4, 2011 at 2:09 PM, Mike Haun <[email protected] >> <mailto:[email protected]>> wrote: >> > I have Sipx 4.2.1 running for a client and they've been very >> happy for about >> > a year. >> > I just switched sip providers from A to B. The only thing I had >> to change >> > in my sipx configs was the IP address of of the SIP Trunk gateway. >> > >> > Here's the odd thing: >> > >> > Test 1 >> > A call comes in and finds it's path in a dial plan which is a >> custom type DP >> > that has an extension number (6302) in the "Resulting Call" >> field. Ext >> > 6302 does not have a phone but it's Call Forwarding parameters >> are set to >> > ring first for 1 second, then if no response forward to <an outside >> > number>. It forwards the call just fine, but there is NO AUDIO >> in either >> > direction! >> > >> > Test 2 >> > Now, another call comes in and it's dial plan sends it to an >> AutoAttendent >> > and since the caller knows the extension number they dial 6302 >> which sends >> > the call to extension 6302 which forwards to <an outside number> >> as in Test >> > 1, but using this method it is a crystal clear connection with >> both way >> > audio! >> > >> > I've looked everywhere within Sipx configs to see if I had two >> places where >> > the IP address needed changed, but I only found the one place at >> Devices -> >> > Gateways -> Sip Trunk Gateway -> Address. >> > >> > I cannot figure this out and am hoping you will provide me with some >> > guidance. >> > >> > Thank you. >> > >> > _ >> >
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