[asterisk-users] Asterisk - VoiceGenie IVR

[Eric Rousse]

Hi,

I'm currently working on a setup between Asterisk and VoiceGenie (which 
is a IVR system).

The way my setup is done, is that I have a PRI line coming in my 
Asterisk server, and then VoiceGenie is connected to Asterisk via SIP, 
like any other softphone basically. I'm able to receive calls in 
Asterisk and then link them with VoiceGenie. But one of my issues is 
that when I get an outside call, transfer the call to VoiceGenie, then 
for that specific calls VoiceGenie would decide that this call has to be 
transfered to an outside party, so then VoiceGenie calls up that number, 
it goes through Asterisk and it reached the other person. But the link 
doesn't stay up very long, max 15 seconds.

That's one of the errors that I see in Asterisk(for obvious reasons I've 
replaced some numbers with *):
-- Hungup 'Zap/8-1'
Feb 15 14:10:19 WARNING[25664]: chan_sip.c:1227 retrans_pkt: Maximum 
retries exceeded on transmission 
7E760C00-C443-6205-772B-88682E2484DE-5060 at 10.1.1.40 for seqno 1 
(Critical Response)
Feb 15 14:10:28 WARNING[25664]: chan_sip.c:1227 retrans_pkt: Maximum 
retries exceeded on transmission 
7E760C00-C443-6205-772B-88682E2484DE-5060 at 10.1.1.40 for seqno 1 
(Critical Response)
   -- Hungup 'Zap/1-1'


Here's a part of my dialplan for outside calls:
exten => _9XXXXXXXXXX,1,Set(CALLERID(all)=<450-655-****>)
exten => _9XXXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

And here's a Macro that I use for incoming call for VoiceGenie:
[macro-voicegenie]
exten => s,1,Answer
exten => s,2,SIPAddHeader(X-Asterisk-DID: ${ARG1})
exten => s,3,SIPAddHeader(X-Asterisk-CallerName: ${ARG2})
exten => s,4,Dial(SIP/108)

exten => 514380****,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten => 514380****,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten => 514373****,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten => 514373****,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten => 514373****,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})


Here's the config in sip.conf:
[108]
type=friend
context=internal
host=10.1.1.40
callerid=VoiceGenie <108>
progressinband=never
disallow=all
allow=ulaw


Also, the support team at Voicegenie they asked me if I stop sending 
"183 Session Progress" before "180 Ringing".
It seems that this could be part of my issue.

Thanks,

-- 
Eric Rousse
System Administrator
514.380.2992
450.655.1001
1.888.641.5800

Telmatik inc.
204 Montarville, suite 250
Boucherville, QC, Canada
J4B 6S2

www.telmatik.com