Manual Dialling Fine. Auto is not.

[bigbozza]

Hi guys, i've got a very odd problem that i'm hoping can be fix rather than just getting a new provider. As mentioned in the title, agents can make manual calls but not auto. As in manaually dialled the numbers from their xlite soft phones work but the call conference/handover of an automatic campaign never happens.

2 different vicidial installations, 2 different providers

both servers work with old provider - even Double NAT'd over the internet
NO servers work with new provider for auto-dialling (but fine for manual calls)

New Server:
Vicidial REDUX VERSION: 2.10-449a, BUILD: 140817-0928
openSUSE 13.1
Asterisk 1.8.29.0-vici
New Dell Server Xeon blah blah <---- it's not really relevant here

Old Server:
Vicidial VERSION: 2.2.1-237, BUILD: 100510-2015
Debian Squeeze 6.0.5
Asterisk 1.4
IBM Server

New provider is Globe Telecom (an small Australian reseller of M2, Commander, Primus). They have provided an EFM link with a WAN IP and they allow unauthenticated calls through to their server via that link. After making many enquiries and being passed through to the top tech for voice at M2, he tells me that he is sure the problem is related to a 180 ringing or 183 - but couldn't be more specific.

WHAT I HAVE TRIED:
Adding Progress() to the dial plan (seems to just hangup the calls automatically)
prematuremedia=yes
nat yes and no
progressinband = yes and no
codecs
canreinvite yes and no (although that doesn't look like it was ever going to help)
Different variations of tTor at the end and even ,60,tTor
Many other things I just can't remember atm - i've been at this for 4 days straight

I've tried to plead with the powers that be to just stay with the previous provider (Neural Networks) as they have been really helpful with asterisk issues in the past, but the almighty dollar wins out in the end I guess.

This board won't let me post a CLI output because I am too new apparently - so if someone could kindly reply to me (with anything), I think it will let me paste.

I will post carrier details however:

[RENAMED]
allow=ulaw
type=friend
insecure=invite,port
host=203.134.5.10
dtmfmode=rfc2833
context=trunkinbound
jitterbuffer=yes
forcejitterbuffer=yes


Dial plan (As you will see in the logs it's clearly dialing Australian numbers so i'm sure that is fine):
exten => _[42378]XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _[42378]XXXXXXXX,2,Monitor(wav,${EXTEN}_${STRFTIME(,,%F_%H%M%S)},m)
exten => _[42378]XXXXXXXX,3,Dial(${GLOBE}/${EXTEN},60,tTor)
exten => _[42378]XXXXXXXX,4,Hangup

exten => _0[42378]XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _0[42378]XXXXXXXX,2,Monitor(wav,${EXTEN}_${STRFTIME(,,%F_%H%M%S)},m)
exten => _0[42378]XXXXXXXX,3,Dial(${GLOBE}/${EXTEN},60,tTor)
exten => _0[42378]XXXXXXXX,4,Hangup

[bigbozza]

It's complaining about too many URLS now so here is a pastebin:

pastebin . com / VFbaGJ4i - remove spaces

[bigbozza]

Final bump before getting paid support

[geoff3dmg]

I've had this problem in the past with Orbtalk in the UK. According to what matt told me at the time it was due to the carrier introducing a delay between the call picking up and the rtp audio starting. I fixed it by adding some extra sip-silence lines in the routing extension.

Relevant threads:

viewtopic.php?f=4&t=32450&p=109242&hilit=orbtalk#p109242
viewtopic.php?f=8&t=26861&p=94225&hilit=orbtalk#p94225

[bigbozza]

Thank you Geoff. In the end I needed to add 3 sip silences in total to extensions.conf

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,n,Playback(sip-silence)
exten => 8368,n,Playback(sip-silence)
exten => 8368,n,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,Hangup()

I hope this helps someone someday.