SIP CONFIG:

[ruben23]

hi guys anyone have idea on this: the VOIP provider have given this sip config and extension setting for the voip trunk: how do i setup this setting on my vicidial version 2.0.5

Is this a valid config im in doubt but i already double check it-they tell thats really there config.

http://pastebin.com/m7d2b938b

[williamconley]

looks reasonable. have you tried it?


obviously it will need modification to match our context and keep our dial switches.

allow=all
canreinvite=no
context=from-trunk
fromdomain=99001315.grnvoip.com
host=99001315.grnvoip.com
outboundproxy=sbc.grnvoip.com
insecure=port,invite
type=peer

on extensions.conf:

[from-trunk]
exten => _Z.,1,Dial(SIP/70.42.72.49/9900131501${EXTEN})

[ruben23]

i got this error:

- Executing AGI("SIP/cc110-b7a017d0", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc110-b7a017d0", "SIP/SIPtrunk/12127775678||To") in new stack
-- Called SIPtrunk/12127775678
Jun 15 21:03:50 NOTICE[3121]: chan_sip.c:9879 handle_response_invite: Failed to authenticate on INVITE to '"cc110" <sip:cc110@9901315.grnvoip.com>;tag=as63d0bac5'
-- SIP/SIPtrunk-08c5b308 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("SIP/cc110-b7a017d0", "") in new stack
== Spawn extension (default, 912127775678, 3) exited non-zero on 'SIP/cc110-b7a017d0'
-- Executing DeadAGI("SIP/cc110-b7a017d0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ESTION----



upon the call..

[williamconley]

looks like you may have put some things in the wrong places. what did you put where in the carrier setup?

[ruben23]

williamconley:

on my sip.conf

i put this

[SIPtrunk]
allow=all
canreinvite=no
context=from-trunk
fromdomain=99001315.grnvoip.com
host=99001315.grnvoip.com
outboundproxy=sbc.grnvoip.com
insecure=port,invite
type=peer
qualify=1000

and on the extensions.conf - i put this

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=SIP/SIPtrunk ; Trunk interface
TRUNKX=SIP/SIPtrunk ; 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
TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569 ; used for blind monitoring
TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for testing

#include extensions-vicidial.conf


thats all i made... what could be the problem.

[williamconley]

did your agent log in and hear "you are the only person in this conference"?

[ruben23]

im testing the call on a softphones directly, after the softphones and the IS register- i dial 91 + the number then have that error log on the CLI.

[williamconley]

exten => _Z.,1,Dial(SIP/70.42.72.49/9900131501${EXTEN})

in that case you will need to modify the "Dial" command to use this method for this carrier, but keep your switches.

they are looking for SIP/70.42.72.49/9900131501NXXNXXXXXX

and you are sending SIP/9901315.grnvoip.com/1NXXNXXXXXX

so you'll need to manipulate the "dial" command slightly. but remember to keep the switches!

[ruben23]

hmmm.. how would i do that, ill add up a new context to the dial plan like this:

[from-trunk]
exten => _Z.,1,Dial(SIP/70.42.72.49/9900131501${EXTEN})

would this not affect the whole setting of the dial plan im using 2.0.5 version.

[williamconley]

you CHANGE the "dial" line, but you don't delete the other lines and you don't delete the switches that come at the end of the dial command.

your provider doesn't care about the scripts that run before or after. so the line number doens't need to change. only the actual contents of the "dial" command itself are seen by the provider. and your provider should have no problem with the switches being there.

it's also possible that your provider was rejecting the call for some OTHER reason (perhaps they dont accept calls with no valid caller id, and your manually dialed call didn't have one, that sort of thing).

so for TESTING purpose, you could attempt to use their "dial" command verbatim (just for the test call to see if you CAN make a call). obviously if that doesn't work, it's time to contact the provider and see what's up.

after that works you can work on modifying it to get the switches in place and the scripts before and after if you took them out.

[ruben23]

i was not able to successfully set the dial plan after a couple of try still i cant get through the calls,

If i modify the dial plan- on what part should i do it_im maybe modifying the wrong part.

and what you mean switches, hope just alittle guide so i can push through with this , im confuse.

[williamconley]

your first step should be to get the dial plan working exactly per the instructions from your VOIP provider. at this point you don't even know if it works.

call them, work through whatever you need to "get it going in asterisk".

AFTER that, you can modify it to work in Vicidial.

"switches" are the ,tTo, |tTo| portion of the dial command (that part of the "comment" in extensions.conf where it says the "o" switch is very true ... the t and T have special purpose and are optional, read up on the asterisk dial command and you will find out).

but before you do any of that (except the reading up part, which is highly recommended), get with your provider and ask exactly how to set up asterisk for a call to work through them. AFTER that trunk works, you can configure vicidial to use it.

[ruben23]

yeah i called my voip provider and they tell me that there config for asterisk is this:

[grnvoip]
allow=all
canreinvite=no
context=from-trunk
fromdomain=99001315.grnvoip.com
host=99001315.grnvoip.com
outboundproxy=70.42.72.140
insecure=very
type=peer


on extensions.conf:

[from-trunk]
exten => _Z.,1,Dial(SIP/70.42.72.140/9900131501${EXTEN})

and said that:

the point is to point it to 70.42.72.140
with this premium prefix - 990131501

and aside that we test it with local softphones without dialer it working fine.

Now would be the part modifying the dial plan of the vicidialnow.
which would be the hardest part.

[williamconley]

Now would be the part modifying the dial plan of the vicidialnow.

it's GOOD that your soft phone dialed, that should make it very possible to do. but until asterisk can duplicate the feat ... you could be looking at a "nevergonnawork" scenario for some hidden reason.

get it working in ASTERISK first. work with the provider to get that done. forget vicidial. after ASTERISK can make a call (using the console to dial or using your softphone to dial through asterisk) ... THEN try vicidial.

if you can't get it to dial in asterisk, you'll never get it to work in vicidial. (no joke).

[ruben23]

hi, yes im focusing now on the asterisk server,

i acquired anew voip telco with much less complicated setting: now

i can dial by soft phones passing through to the asterisk dialer can hear clients and they can hear me., but when i tried using the vicidial-with predicitve dialing having ratio 1:2, I get Livecall but can hear the client side- i know that the client is responding coz the softphones voice indicator are moving-( means someone is talking)

then now befroe that when i register my softphones then login with my phone id on the vicidialer during welcome rings and i answered i cant hear the voice conferece( saying your the only person in this conference) and time i click resume thats the time the conference voice would voice out- with 10 callers testing i got only 2 able to hear it

but all agents are log in the conference and can dial but cant hear the client side. this is my ouptut on the logs:

http://pastebin.com/m56d9bcb7

what could be the problem: im using

vicidialnow ce 1.2

[williamconley]

this would be something for which you should open a whole new thread.

support for soft phones and sound cards on your workstations is what you seem to be asking now ... but i'm not sure. (if the indicators on your screen for the soft phones show sound, but the sound doesn't come out the earpiece/speaker ... that's not an asterisk issue ...)