vicidial.org

Hi Guys

VICIDIAL V8.1, VERSION: 2.14-717a, BUILD: 190724-1603, Single Server Install on HP Proliant ML Gen 10 Octa Core Xeon Server with 16GB RAM and 1.5TB SSD along with this i have one Dinstar DWG Series gsm gateway and another two DInstar UC Series gsm Gateways. My location is India and we are more of an outbound call center.

I have done the installations and configurations and everything was ok till i wanted to add this second and third gsm gateways. I made a new trunk in sip.conf file and also added an entry in the extensions.conf file but still it will not make calls via the second gateway. I do not know what i am doing wrong. Can someone please guide me

MY sip.conf file looks like this:-

[gsm222]
username=gsm222
secret=dinstar
type=friend
allow=all
qualify=yes
host=dynamic
nat=yes
dtmfmode=rfc2833
context=trunkinbound

[gsm333]
username=gsm333
secret=dinstar
type=friend
allow=all
qualify=yes
host=dynamic
nat=yes
dtmfmode=rfc2833
context=trunkinbound

My extensions.conf looks like this

[general]
static=yes
writeprotect=no

[globals]

CONSOLE=Console/dsp ; Console interface for demo
TRUNK=DAHDI/r1 ; Trunk interface
TRUNKX=DAHDI/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface
SIPtrunk=SIP/1234:PASSWORD@sip.provider.net ; SIP trunk

#include extensions-vicidial.conf

[trunkinbound]
exten => _X.,1,AGI(agi-DID_route.agi)
exten => _X.,n,Hangup()
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}-----${HANGUPCAUSE(${HANGUPCAUSE_KEYS()},tech)}))

[loopback-no-log]
exten => _999XX11112,1,Wait(2)
exten => _999XX11112,n,Answer()
exten => _999XX11112,n,Playback(ss-noservice)
exten => _999XX11112,n,Playback(vm-goodbye)
exten => _999XX11112,n,Hangup()

[default]
include => vicidial-auto

exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,2,Dial(SIP/gsm222/${EXTEN:1},,tToR)
exten => _9X.,3,Hangup()

exten => _8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8X.,2,Dial(SIP/gsm333/${EXTEN:1},,tToR)
exten => _8X.,3,Hangup()

And i have set the dial prefix to 8 in two campaigns, I have a total of 4 campaigns.

The issues is that the calls do not go through the second gateway, if i try to make calls it just gives me an error like Call Rejected: CHANUNAVAIL Cause: 20 - Subscriber absent

Can someone please guide me over here it would be really helpful.

Thanks a lot in advance.

post the asterisk cli output from a GOOD and from a BAD call attempt. Two calls, start to end. Not 3000 lines of unrelated code, but don't strip off any start or end lines from the calls in question. Of course you can change the last four or five digits of any phone numbers to Zeros, but leave the first few digits to preserve the dial pattern for matching purposes.

Hi
Thanks william for the response but i sorted this one out, actually i had made some additional changes in the extensions.conf file and now it is working. Thanks for helping out.

Advice: Post your solution here. Both for posterity (for you) so you don't lose the fix six months from now after your notes are otherwise gone AND because the next guy may in fact benefit from your solution.

I agree with Sir William,, you might think it the solution but then in the end it's just the bandaid of the problem..

Hi sorry for the delayed reply

just forgot to do dialplan reload

after that it was working, but then recordings were not happening for conversations going through that particular gateway so i just modified the extensions.conf file to something like this:-

[general]
static=yes
writeprotect=no

[globals]

CONSOLE=Console/dsp ; Console interface for demo
TRUNK=DAHDI/r1 ; Trunk interface
TRUNKX=DAHDI/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface
SIPtrunk=SIP/1234:PASSWORD@sip.provider.net ; SIP trunk

#include extensions-vicidial.conf

[trunkinbound]
exten => _X.,1,AGI(agi-DID_route.agi)
exten => _X.,n,Hangup()
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}-----${HANGUPCAUSE(${HANGUPCAUSE_KEYS()},tech)}))

[loopback-no-log]
exten => _999XX11112,1,Wait(2)
exten => _999XX11112,n,Answer()
exten => _999XX11112,n,Playback(ss-noservice)
exten => _999XX11112,n,Playback(vm-goodbye)
exten => _999XX11112,n,Hangup()

[default]
include => vicidial-auto

exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,n,Dial(SIP/gsm222/${EXTEN:1},,tToR)
exten => _9X.,n,Dial(SIP/gsm333/${EXTEN:1},,tToR)
exten => _9X.,n,Hangup()

and now it is working.

Thanks for helping out and hope that this helps someone. Cheers