I was able to find the answer. By using the Answer with a priority of 1 before the Dial, the call signalling is setup properly.
On Sat Jan 29 15:51:22 PST 2005, [EMAIL PROTECTED] wrote:
Asterisk experts,
I have been pulling my hair out troubleshooting what appears to be a Zap channel disconnect on the TDM side. I am trying to have an inbound call on a T1 zap channel on the TE405P from our Harris switch automatically dial a SIP channel. Any help appreciated very appreciated.
When the call in extensions.conf executes the demo-congrats: exten => 2145551234,1,playback(demo-congrats)
I hear the demo - success.
When the call in extensions.conf tries to automatically dial a SIP channel:
exten => 2145551234,1,Dial(SIP/100)
Asterisk sends an IVITE to the phone followed by an immediate CANCEL. Asterisk console debug looks like the Zap Channel is hanging up.
So my question is: Is this how inbound calls from a switch need to automatically dial to a SIP channel, or does the call need to be answered in some other way, and then execute the Dial command? Does anyone please have any pointers, or example configuration for this kind of call scenario?
It is as if the signalling is not correct and the Harris is not getting some kind of wink signal when I try to Dial SIP, and the Harris hangs up the call, but when the demo playback executes, the Harris gets the right signal. Can Asterisk actually debug the Zap channel to see where the call hangup is coming from, and if so, how?
Or maybe my zapata / zaptel configuration is not correct for this kind of call scenario. Here is the relevant config:
zapata.conf:
context=Provider_T14
signalling=sf_featdmf
group=4
channel => 73-96
zaptel.conf: span=1,0,0,esf,b8zs #span=2,1,0,esf,b8zs span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs
e&m=1-24 e&m=25-48 e&m=49-72 e&m=73-96
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