Call Rejected: CHANUNAVAIL Cause: 20 - Subscriber absent

[Princeamor]

Hello I am able to log in and hear the message that says that I am the only person in the conference and I hear the boop sound when I click for a call but I don't hear a ring or the prospect. Prospect information is shown. On the top of the screen it says waiting for ring... (x) seconds. I have already reinstalled VicidialNOW on the server but that did not help. I checked my vitelity account balance and that is ok. I was able to make calls previously but then things started screwing up when I upgraded to v2.2 and after that I reinstalled 2.0 and now I am having this problem. Does anyone know what could be the cause of this and how I could fix it?

Thanks for your suggestions...


Specs:
VicidialNOW VERSION: 2.0.5-206
Twinkle
Asus p5kc mobo
Intel Core 2 Quad 6600
2GB RAM
Western Digital SATA HD

No additional hardware.

[Princeamor]

bump

[williamconley]

too bad you bumped it yourself, i would have seen this two days ago. but i missed it because it looked like someone answered you (two entries usually means one question, one answer). and i use the "unanswered" button to check routinely.

oh, well.

when you post, please post your entire configuration including (but not limited to) your installation method and OS with kernel or version, vicidial version and build, asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)
_______

Asterisk CLI output during your issue (along with the above information that's missing ...)?

[Princeamor]

Specs:
VicidialNOW VERSION: 2.0.5-206
BUILD: 90522-0506
Asterisk (?)
Twinkle
Asus p5kc mobo
Intel Core 2 Quad 6600
2GB RAM
Western Digital SATA HD
No additional hardware
Single Server
No Digium/Sangoma Hardware
No Extra Software After Installation

[williamconley]

Code: Select all

asterisk -R

will provide your asterisk version

[Princeamor]

Hello I re-downloaded and burned Goautodial iso CE 2.0 RC2 and it installed successfully however I still have the same issue. Now it gives me this error when I dial for customer:

DIAL ALERT:

Call Rejected: CHANUNAVAIL
Cause: 20 - Subscriber absent

Here are my updated specs:

Vicidial VERSION: 2.2.1-237
BUILD: 100510-2015
Asterisk 1.4.27.1-1
Twinkle
Asus p5kc mobo
Intel Core 2 Quad 6600
2GB RAM
Western Digital SATA HD
No additional hardware
Single Server
No Digium/Sangoma Hardware
No Extra Software After Installation

(btw my screen looks nothing like the goautodial getting started guide found on this page: http://goautodial.com/wiki/getting-started-guide/)

Thank you very much williamconley for all your help

[williamconley]

1) Where exactly do you see this message? on your soft phone? agent screen?

2) asterisk CLI during the event? (not 3000 lines of code, just the specific moment of the attempted call, if there is NO activity use sip debug and see if there is any activity with that on)

3) is your phone a SIP/soft phone? is it registered to the server?

4) what kind of trunk are you trying to use?

Asterisk CLI:

Code: Select all

asterisk -R

SIP debug inside Asterisk CLI:

Code: Select all

sip debug

[Princeamor]

I see this message on the agent screen after waiting for a few seconds for a ring when attempting to make a call.

asterisk -r gives me this message:
WARNING: chan_sip.c:3095 create addr: No such host: inbound23.vitelity.net

and other similar errors. I used sip set debug however the same warning messages keep appearing. I also changed the carrier info according to the vitelity website instead of the previous vicidialnow walk-through guide and continued to get the same error messages.

Yes my phone is a soft-phone (twinkle) and I use un:cc100 pw:test and it gives me a registration succeeded message when connected to the server.

Not sure about the trunk.

Now whenever I try to Dial my customer information says undefined in every entry.

Any suggestions?

[williamconley]

please list your information for the carrier from admin->carriers for this carrier specifically (do not post your user/username/secret/passwords, just replace those with xxxx)

you may want to try to go over that in your instructions again and/or use one of the samples more closely

[Princeamor]

I also get this Error message after typing asterisk -r: ERROR[5896]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe.

Also, the customer information on the agent screen when attempting to make a call only says undefined.

Try 1:

Registration String

register=>xxxxx:xxxxx@inbound23.vitelity.net


Account Entry:

[vitel-outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
context=outbound
username=xxxxx
fromuser=xxxxx
trustrpid=yes
sendrpid=yes
secret=xxxxx
allow=all
nat=yes


Dialplan Entry:

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@vitel-outbound,,o)
exten => _91NXXNXXXXXX,3,Hangup


Try 2:

Registration String

register => xxxxx:xxxxx@inbound24.vitelity.net:5060

Account Entry:

[vitel-inbound]
type=friend
dtmfmode=auto
host=inbound24.vitelity.net
context=inbound
username=xxxxx
secret=xxxxx
allow=all
insecure=very
nat=yes

[vitel-outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
context=outbound
username=xxxxx
fromuser=xxxxx
trustrpid=yes
sendrpid=yes
secret=xxxxx
allow=all
nat=yes

Dialplan Entry:

[outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/$\{EXTEN}@vitel-outbound)
exten => _011.,1,Dial(SIP/$\{EXTEN}@vitel-outbound)

; e911 must be enabled. see DIDs > NPANXXNXXX > Action > e911
exten => _911,1,Dial(SIP/911@vitel-outbound)

Try 3

Registration String

register => xxxxx:xxxxx@inbound24.vitelity.net:5060

Account Entry:

[vitel-outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
context=outbound
username=xxxxx
fromuser=xxxxx
trustrpid=yes
sendrpid=yes
secret=xxxxx
allow=all
nat=yes

Dialplan Entry:

[outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/$\{EXTEN}@vitel-outbound)
exten => _011.,1,Dial(SIP/$\{EXTEN}@vitel-outbound)

; e911 must be enabled. see DIDs > NPANXXNXXX > Action > e911
exten => _911,1,Dial(SIP/911@vitel-outbound)

[williamconley]

1) ast_carefulwrite: write() returned error: Broken pipe is normal (it's a message stating that it didn't write anything in the log because there was nothing to write, but the write function hates it when you don't write anything and tosses an error)

2) what directions are you following for setting up your account entry and dial plan entries for your carrier? that is where your problem is ... I see three tries, but where are they coming from?

Create a single carrier with two contexts (one for inbound and one for outbound) where the only difference between the contexts is the context name ([vitel-inbound] vs [vitel-outbound]) and the host (provided by vitelity). The "context=" line in both should be "context=trunkinbound" because that line is ONLY necessary for inbound anyway and all calls inbound should be directed to trunkinbound for VICIdial.

Then USE the global variable to convert your "dial plan" trunk into your actual trunk (makes your dial plan easier to modify). Look at the samples already in the system.

Code: Select all

DIAL9TRUNK=SIP/vitel-outbound

assuming the context for your outbound vitelity account is [vitel-outbound]

The use a "standard" dial 9 dial plan (dial plan entries do not have a [context] at the top of them, they are included in the default extensions.conf context and that breaks it away so it can no longer be found in default):

Code: Select all

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${DIAL9TRUNK}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup

Then have a look at your asterisk CLI and see if it attempts to dial out through vitelity when you make a call (try manual dialing from a phone and/or within a campaign first to test the system).

[gardo]

Posting the exact output of your Asterisk CLI when dialing might help a lot.

[williamconley]

yep. i was gonna go there next, but i didn't want to "overwhelm"

i also wanted to toss in a warning about "NOT" 3000 lines of code, just include the part that happens during a dial attempt.