Dialplans.

[ndimension]

Hi.

First of all. Laymans way of doing vici:

I work in Poland. To call a local landline using a landline I dial 817479546 where the 81(or 82, 83 etc) is area code. If someone in ie UK where to call me he would have to dial 48817479546, where the 48 is Poland code, 81 is my area code, and rest is my number.

This is taken from directly from carrier configuration page:

register => 0814708115:xxxxxxx@80.249.5.196:5060/0814708115

[goautodial]
disallow=all
allow=g729
type=friend
secret=xxxxxx
username=0814708115
host=80.249.5.196
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
fromdomain=80.249.5.196



exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@goautodial,,tTo)
exten => _91XXXXXXXXXX,3,Hangup


I get this:

Call Rejected: CHANUNAVAIL
Cause: 20 - Subscriber absent.

No inbound, no outbound.

Help?


Vicidial 2.2.1
Asterisk 1.4.27.1-vici
Dahdi 2.3
VtigerCRM 5.1.0
Sangoma 3.5.14
CentOS 5.5

Single Server
No Digium/Sangoma Hardware
No Extra Software After Installation

[striker]

hi

paste one single full call asterisk cli result here

[ndimension]

Hmm......
I see that vici is gonna be more fun than I've expected

hi

paste one single full call asterisk cli result here

How do i do that?

[striker]

in ssh terminal type asterisk -vvvvvvr


now make a call from your extension

you can see the call flow in that screen
copy that result and paste it here


if you no idea about ssh teminal

in google type putty , download it , enter your server ip select ssh

and press enter it will ask for username and password

[williamconley]

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@goautodial,,tTo)
exten => _91XXXXXXXXXX,3,Hangup

this is a US dial pattern. probably ought to adjust it for poland ...?

Also: try using a different provider (goautodial's upline may not support poland, but you may want to consult with gardo on that)

[ndimension]

cli

Also, the lady says that she's sorry ---- but thats not a valid extension.


== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.27.1-1 RPM by demian@goautodial.com currently running on go (pid = 2376)
Verbosity is at least 21
[Apr 27 12:02:04] == Parsing '/etc/asterisk/manager.conf': [Apr 27 12:02:04] Found
[Apr 27 12:02:04] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 27 12:02:04] == Parsing '/etc/asterisk/manager.conf': [Apr 27 12:02:04] Found
[Apr 27 12:02:04] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 27 12:02:04] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 27 12:02:09] == Parsing '/etc/asterisk/manager.conf': [Apr 27 12:02:09] Found
[Apr 27 12:02:09] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 27 12:02:09] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 27 12:02:10] ERROR[9316]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Apr 27 12:02:10] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 27 12:02:18] == Parsing '/etc/asterisk/manager.conf': [Apr 27 12:02:18] Found
[Apr 27 12:02:18] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 27 12:02:18] -- Executing [8600052@default:1] MeetMe("Local/8600052@default-9002,2", "8600052|F") in new stack
[Apr 27 12:02:18] > Channel Local/8600052@default-9002,1 was answered.
[Apr 27 12:02:18] == Starting Local/8600052@default-9002,1 at default,91817479546,1 failed so falling back to exten 's'
[Apr 27 12:02:18] == Starting Local/8600052@default-9002,1 at default,s,1 still failed so falling back to context 'default'
[Apr 27 12:02:18] -- Sent into invalid extension 's' in context 'default' on Local/8600052@default-9002,1
[Apr 27 12:02:18] -- Executing [i@default:1] Playback("Local/8600052@default-9002,1", "invalid") in new stack
[Apr 27 12:02:18] -- <Local/8600052@default-9002,1> Playing 'invalid' (language 'en')
[Apr 27 12:02:18] WARNING[9374]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
[Apr 27 12:02:18] WARNING[9374]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
[Apr 27 12:02:22] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 27 12:02:22] == Auto fallthrough, channel 'Local/8600052@default-9002,1' status is 'UNKNOWN'
[Apr 27 12:02:22] -- Executing [h@default:1] DeadAGI("Local/8600052@default-9002,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 27 12:02:23] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 27 12:02:23] == Spawn extension (default, 8600052, 1) exited non-zero on 'Local/8600052@default-9002,2'
[Apr 27 12:02:23] -- Executing [h@default:1] DeadAGI("Local/8600052@default-9002,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 27 12:02:23] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 27 12:03:04] == Parsing '/etc/asterisk/manager.conf': [Apr 27 12:03:04] Found
[Apr 27 12:03:04] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 27 12:03:04] == Parsing '/etc/asterisk/manager.conf': [Apr 27 12:03:04] Found
[Apr 27 12:03:04] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 27 12:03:04] ERROR[9567]: utils.c:966 ast_carefulwrite: write() returned error: Connection reset by peer
[Apr 27 12:03:04] ERROR[9567]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Apr 27 12:03:04] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 27 12:03:09] == Parsing '/etc/asterisk/manager.conf': [Apr 27 12:03:09] Found
[Apr 27 12:03:09] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 27 12:03:10] ERROR[9588]: utils.c:966 ast_carefulwrite: write() returned error: Connection reset by peer
[Apr 27 12:03:10] ERROR[9588]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Apr 27 12:03:10] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 27 12:03:10] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 27 12:04:04] == Parsing '/etc/asterisk/manager.conf': [Apr 27 12:04:04] Found
[Apr 27 12:04:04] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 27 12:04:04] == Parsing '/etc/asterisk/manager.conf': [Apr 27 12:04:04] Found
[Apr 27 12:04:04] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 27 12:04:04] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 27 12:04:09] == Parsing '/etc/asterisk/manager.conf': [Apr 27 12:04:09] Found
[Apr 27 12:04:09] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 27 12:04:09] ERROR[9797]: utils.c:966 ast_carefulwrite: write() returned error: Connection reset by peer
[Apr 27 12:04:09] ERROR[9797]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Apr 27 12:04:09] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 27 12:04:10] ERROR[9774]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Apr 27 12:04:10] == Manager 'sendcron' logged off from 127.0.0.1




Does the black magic above means that i should remove 91 (it should be - for local calls 81) from dialplan and put XX/81 in its place?

[williamconley]

Also, the lady says that she's sorry ---- but thats not a valid extension.

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@goautodial,,tTo)
exten => _91XXXXXXXXXX,3,Hangup

this is a US dial pattern. probably ought to adjust it for poland ...?

Your numbers must match the pattern ... "_" (underscore) indicates that it is a pattern, 9 and 1 must match exactly and X will match any number. Adjust this to match YOUR pattern. The :1 in EXTEN:1 will "strip off" the first digit (so this "9" as a dial prefix on the campaign will choose THIS carrier, then be discarded, which is a standard method of allowing multiple carriers).

If you have no requirement to dial a 0 or 1 before your phone numbers, you can change to _9XXXXXXXXX (the number of X's will be the length of your phone numbers) then the EXTEN:1 will strip off the 9 and your number will be transmitted to your provider. You may need to choose "omit phone code" in your campaign if you do not need to dial the 0 or 1, as I believe vicidial forces a phone code during upload.

Area code is considered part of the phone number, and should only be specified if you need to use a different carrier based on the area code being dialed (ie: use X to represent each digit in the area code unless you intend to make a separate extension for each area code to allow you to point each area code to a specific carrier)

[striker]

just the replace the following in your carrier setting with one i listed below

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@goautodial,,tTo)
exten => _91XXXXXXXXXX,3,Hangup

replace with

exten => _8XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8XXXXXXXX,2,Dial(SIP/${EXTEN}@goautodial,,tTo)
exten => _8XXXXXXXX,3,Hangup

and then dial the 817479546 from the one of the extensions, no need to login as a agent ,

just register a softphone with an extension and dial 817479546

[williamconley]

IF all your phone numbers start with "8"

[ndimension]

will choose THIS carrier, then be discarded, which is a standard method of allowing multiple carriers.

That part i dont get at all

EXTEN:1 will strip off the 9 and your number will be transmitted to your provider.

Hookay - What if i want to remove the EXTEN:1 part from dialplan?

replace with

exten => _8XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8XXXXXXXX,2,Dial(SIP/${EXTEN}@goautodial,,tTo)
exten => _8XXXXXXXX,3,Hangup

If i do have a requirement to dial 0 before my phone number - I should add 0 before as in:

_08XXXXXXXX

Right?
It doesnt work btw - still not a valid extension.

Want cli again?

separate extension for each area code to allow you to point each area code to a specific carrier

That means i can call one area code with this voip provider and another area code with another provider. Extension is a way of telling server "use this here and that there". Am i close?

Update:

My voip provider says i need 0817479546 format where the:

- 0 is a must for all numbers
- 81 is my area code
- 7479546 is number im trying to call

I've tried these do far and "no go"

_XXXXXXXXX
_081XXXXXXX
_08XXXXXXXX

[ndimension]

Check this:


http://en.wikipedia.org/wiki/Telephone_numbers_in_Poland


My list (testlist101.csv) has numbers in 0817479546 format
So if i do ------------------------------->_XXXXXXXXX it should work.

When i upload csv file i input:
"list id override" as 101
"phone code override" as 1

I crated a new list ("Add a new list"), gave it 101 id, set active to Y. That removed my problem with "no leads in hopper".

I see that there's a connection in dialplans' (_xxxxxx etc), phone_number format in csv file and probably "phone code override"


I'll try adding 48 now.
Edit: 48 does nothing.

I'm a patient doode most of time, but I'm starting to look more keenly on my fathers 10 pound hammer after every dialplan_change.

Help?

[williamconley]

show your present dialplan and CLI output from a failed call (not 3000 lines of code please, just from "dial" to "hangup" on that one call )

[ndimension]

exten => _XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXX,2,Dial(SIP/${EXTEN}@goautodial,,tTo)
exten => _XXXXXXXXXX,3,Hangup

[Apr 29 03:34:48] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 29 03:34:48] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-c93f,2", "8600051|F") in new stack
[Apr 29 03:34:48] > Channel Local/8600051@default-c93f,1 was answered.
[Apr 29 03:34:48] == Starting Local/8600051@default-c93f,1 at default,910817479546,1 failed so falling back to exten 's'
[Apr 29 03:34:48] == Starting Local/8600051@default-c93f,1 at default,s,1 still failed so falling back to context 'default'
[Apr 29 03:34:48] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-c93f,1
[Apr 29 03:34:48] -- Executing [i@default:1] Playback("Local/8600051@default-c93f,1", "invalid") in new stack
[Apr 29 03:34:48] -- <Local/8600051@default-c93f,1> Playing 'invalid' (language 'en')
[Apr 29 03:34:48] WARNING[4109]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
[Apr 29 03:34:48] WARNING[4109]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
[Apr 29 03:34:52] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 29 03:34:52] == Auto fallthrough, channel 'Local/8600051@default-c93f,1' status is 'UNKNOWN'
[Apr 29 03:34:52] -- Executing [h@default:1] DeadAGI("Local/8600051@default-c93f,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 29 03:34:52] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 29 03:34:52] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-c93f,2'
[Apr 29 03:34:52] -- Executing [h@default:1] DeadAGI("Local/8600051@default-c93f,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 29 03:34:52] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[striker]

you are dialling 12 digits (910817479546) and the dialplan you have only accept 10 digits

exten => _XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXX,2,Dial(SIP/${EXTEN}@goautodial,,tTo)
exten => _XXXXXXXXXX,3,Hangup


FYI
X---0 to 9
N---2 to 9
Z--- 1 to 9

{EXTEN:1} will remove the first digit of the number u dialled
{EXTEN:2} wil remove the fisrst two digits of the number u dialled

0${EXTEN} this will add 0 infront of the number u dialled

[williamconley]

Code: Select all

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@goautodial,,tTo)
exten => _91XXXXXXXXXX,3,Hangup

will send 1+10 digits to carrier

Code: Select all

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:2}@goautodial,,tTo)
exten => _91XXXXXXXXXX,3,Hangup

will send just 10 digits to carrier

[ndimension]

This is crazy. I get the part about EXTEN to remove numbers u dont need to get my number. It is working. I'm happy. Then of course i messup some random config and nothing works anymore. So i reinstall, as usual.

I configure as usual, but then again this happens (with comment added by me, mind You, i might be ofc wrong this is by trial and error):

[May 9 05:01:35] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-d7ef,2", "8600051|F") in new stack

Create MeetMe room, "meet" agent logging in "8600051|F" with server "Local/8600051@default-d7ef,2"


[May 9 05:01:35] > Channel Local/8600051@default-d7ef,1 was answered.

That generally says that agent login was successful i think.



[May 9 05:01:35] -- Executing [910817479546@default:1] AGI("Local/8600051@default-d7ef,1", "agi://127.0.0.1:4577/call_log") in new stack

This is where i get the logs, agi whatever it is, logs this very log.



[May 9 05:01:35] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

The logging process is running without errors (return 0 means "ok" i presume)



[May 9 05:01:35] -- Executing [910817479546@default:2] Dial("Local/8600051@default-d7ef,1", "SIP/0817479546@goautodial||tTo") in new stack

Take a lead number fom hopper and execute second command from dialplan - "Dial" with parameters specified.



[May 9 05:01:35] WARNING[12647]: chan_sip.c:3193 sip_call: No audio format found to offer. Cancelling call to 0817479546

This tells me two things. First I messed up some thing.
The second thing it tells me that i got the server to dial the exact number i want it to dial (it cancels call to 0817479546 witch is the correct number to pass to my voip provider servers).




[May 9 05:01:35] -- Couldn't call 0817479546@goautodial

Why does it say @goautodial?



[May 9 05:01:35] == Everyone is busy/congested at this time (0:0/0/0)

Some related "errory" stuff.



[May 9 05:01:35] -- Executing [910817479546@default:3] Hangup("Local/8600051@default-d7ef,1", "") in new stack

Execute the third command from dialplan after you finis doing command 2. "Dial" process is finished (no matter the result) so the server hangups.



[May 9 05:01:35] == Spawn extension (default, 910817479546, 3) exited non-zero on 'Local/8600051@default-d7ef,1'

Some stuff .

[May 9 05:01:35] -- Executing [h@default:1] DeadAGI("Local/8600051@default-d7ef,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack

Well known asterisk error code. It means i fail at this



[May 9 05:01:35] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

Also if for by any astronomically impossible chance i get this working (agents can call) can i post my version of "Vici for dummies with pictures and commentary"?

Edit: Quoted from some other thread

So you dial "9-1-555-555-5555" and the system uses the 9 to choose your trunk, then sends 15555555555 to the SIP/XXXX carrier you defined for that dialplan.

So what exactly does the number 9 do? Is it a normal number to dial (You would dial it on a landline too) or is it a "pointer"-code thing that is interpreted in some way by server.[/quote]

[striker]

[May 9 05:01:35] WARNING[12647]: chan_sip.c:3193 sip_call: No audio format found to offer. Cancelling call to 0817479546


this is your problem

no audio format found

in sip settings check what u have entered for allow= ?

g729 , ulaw or alaw

also check what audio codec does you carrier supports.
then put that format in allow=?

[ndimension]

in sip settings check what u have entered for allow= ?

g729 , ulaw or alaw

also check what audio codec does you carrier supports.
then put that format in allow=?

Provider/carrier supports g711u (preffered) and g729 (worked before). I'll check 711u.


[goautodial]
disallow=all
allow=g729
type=friend
secret=
username=0814708115
host=80.249.5.196
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
fromdomain=80.249.5.196

BTW:

exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:2}@goautodial,,tTo)


What does @goautodial and ,, and tTo do?

[striker]

1. g729 is installed in your server
2. t = time out
3. T = absolute time out
4. o = operator

dont ask further more explanation , i dont no, google it for more information.

goautodial as per your post it is the name of the carrier which ur using

u can see it between [ ] ,
if you give ur name there then below is dialplan
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:2}@ndimension,,tTo)

[williamconley]

2. t = time out
3. T = absolute time out
4. o = operator

um ..

tT=each is to allow either calling party or called party to transfer the call.
o=uses the asterisk v1.0 callerid behavior (required for proper vicidial execution).

But your problem is likely that you have "required" g729 but not installed it on your server. Instead of "allow=g729", try "allow=all" until you have finished testing (and perhaps installed g729).

Yes, the 9 is a "code" to bring the call to this particular carrier. It's not special or magic, though, if your extension is "91X...." (anything starting with a 9), that's where the system knows you meant to use THIS carrier, because you dialed a number starting with 9. The 9 is generated by putting it in the "Dial Prefix" (which could be renamed "Carrier Choice").