Inbound Carrier Settings Confusion

[edmund.uba]

ViciBox_v.7.x86_64-7.0.2.iso | Vicidial 2.12-549a Build 160404-0940 | Asterisk 11.22.0-vici | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel Core2Duo

I am quite confused with the carrier settings of an inbound campaign.
Are these settings correct?

Registration String: register => XXXXXX:XXXXX@XX.XX.XX.XX:5060/XXXXXX


Account Entry:
[welco]
username=XXXXXXX
password=XXXXXXX
disallow=all
allow=alaw
allow=ulaw
type=peer
host=XX.XX.XX.XX
dtmfmode=rfc2833
canreinvite=no
qualify=4000

Dial Plan:
exten => DID-number,1,Ringing
exten => DID-number,2,Wait(1)
exten => DID-number,3,Answer
exten => DID-number,4,AGI(agi://127.0.0.1:4577/call_log)
exten => DID-number,5,Hangup

I tried calling the number but none of the agents are receiving it. Am I doing something wrong with the carrier settings? What should be my dial prefix in the campaign settings?

[Rogger]

Hi,

Follow these steps:

1) Context for incoming calls is "trunkinbound" in your carrier
2) Make sure that you have created DID
3) make sure that you have created INBOUND-Group and select your campaign.
4) Make sure you are getting the digits equal to its DID sent by your carrier.
5) Its basic but don't forget ( asterisk reload or dialplan reload before the tests )

good luck,

Rogger

[Vince-0]

The inbound carrier dial plan doesn't need to be there.

The DID number is picked up from the DIDs in the Inbound menu.

[udy786]

One line missing in your dialplan to route call into your DID setting.


exten => DID-number,1,Ringing
exten => DID-number,2,Wait(1)
exten => DID-number,3,Answer
exten => DID-number,4,AGI(agi://127.0.0.1:4577/call_log)
exten => DID-number,5,AGI(agi-DID_route.agi)
exten => DID-number,6,Hangup

Create a DID on admin page with DID-number and route into your Ingroup.

Make sure you write above dialplan in same context where Inbound call is going. You can see context on CLI if any error getting. Normally it will go into default or trunkinbound.

Thanks
Uday.

[mattyou1985]

have a look at this

viewtopic.php?f=4&t=35922

[edmund.uba]

ok so here is the new settings of the carrier

Registration String: register => XXXXXX:XXXXX@XX.XX.XX.XX:5060/XXXXXX

Account Entry:
[welco]
disallow=all
allow=ulaw
type=peer
username=xxxxxxxx
secret=xxxxxxxxx
host=69.71.197.20
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=port,invite
nat=yes

Dial Plan:
exten => DID-number,1,Ringing
exten => DID-number,2,Wait(1)
exten => DID-number,3,Answer
exten => DID-number,4,AGI(agi://127.0.0.1:4577/call_log)
exten => DID-number,5,AGI(agi-DID_route.agi)
exten => DID-number,6,Hangup

I already have the DID, IN_GROUP, and the agents created, but still I am not receiving calls.

[mattyou1985]

ok so you changed context=trunkinbound as its not defalt we now need to change it in

so now in DID, MODIFY, look for "Extension Context: default " and change it to "Extension Context: trunkinbound " then place a test call it should connect as long as an agent is in wating and has been added to the inbound group

most of this is noted in the manural but we all need help from time to time

[mattyou1985]

One line missing in your dialplan to route call into your DID setting.


exten => DID-number,1,Ringing
exten => DID-number,2,Wait(1)
exten => DID-number,3,Answer
exten => DID-number,4,AGI(agi://127.0.0.1:4577/call_log)
exten => DID-number,5,AGI(agi-DID_route.agi)
exten => DID-number,6,Hangup

Create a DID on admin page with DID-number and route into your Ingroup.

Make sure you write above dialplan in same context where Inbound call is going. You can see context on CLI if any error getting. Normally it will go into default or trunkinbound.

Thanks
Uday.

this is mine

exten => _44.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _44.,2,Dial(sip/4${EXTEN:2}@sipcarriername,,Tto)
exten => _44.,3,Hangup

an i just spoted somthink missing and could allso be rong so looking at his and mine and from what i no yours should look like this

exten => _44.,1,Ringing
exten => _44.,2,Wait(1)
exten => _44.,3,Answer
exten => _44.,4,AGI(agi://127.0.0.1:4577/call_log)
exten => _44.,5,AGI(agi-DID_route.agi)
exten => _44.,6,Dial(sip/44${EXTEN:2}@welco,,Tto)
exten => _44.,7,Hangup

now using this would need a prefix of 44 if you need a difrent prefix that is 2 number long then replace all 44 with your 01 0r 11 or what ever is needed
(prefix)(number)

but if its more than 2 number for prefix change {EXTEN:2} < this will remove the first 2 numbers from diall plan but sip/44$ add's it back in..hope this helps

@sipcarriername << this is defined in account Entry
[welco]

now you should be able to place calls and you should now see the incomming call connecting to your CLI out put

now to see the cli output download http://www.putty.org/

then connect to your PBX using its ip and root name and password then at the

root# typ in asterisk -rvvv

the page will change and at the bottem youl see

CLI# and lots of thigs doing stuff it is hard to read but if you are the onley one on its mush better

place a test call to your inbound you should see the screen change and your test call will show its number in thir somwair

if this is hapning then you are good and can move on from hear but if not then somthink is still rong with

Account Entry: or Dialplan Entry:

to find out if its one or other using putty

root# typ in "asterisk -rvvv"

then

CLI# typ in "sip show peers"

you shouls see your carrier name in thir

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
1111/1111 (Unspecified) D Yes Yes 0 UNKNOWN
4001/4001 192.168.0.55 D Yes Yes 59255 OK (4 ms)
4002/4002 (Unspecified) D Yes Yes 0 UNKNOWN
4003/4003 192.168.0.2 D Yes Yes 58193 OK (3 ms)
4005/4005 (Unspecified) D Yes Yes 0 UNKNOWN
4006/4006 192.168.0.20 D Yes Yes 49072 OK (3 ms)
5555/5555 192.168.0.55 D Yes Yes 59255 OK (8 ms)
gs102/gs102 192.168.0.6 D Yes Yes 54442 OK (5 ms)
welco 192.111.11.111 Yes Yes 5060 OK (83 ms)

if it looks like this your all good and its nothink to do with Account Entry:
and then well try agen on Dialplan Entry: somthink is ither missing or rong by dowing all this you should be able to fix it or at lease see wair its gowing rong

[williamconley]

Inside [trunkinbound] in the extensions.conf file is a "catch-all" extension that sends all calls (regardless of the DID) to the inbound agi script. This script checks the Vicidial database for "what should I do with this call". Nothing else is needed at this point, since control is now in the hands of an agi script using the configuration settings from the Vicidial GUI.

It is not necessary, advantageous, or useful to configure each inbound DID in any context in extensions.conf ([trunkinbound],[default], or anywhere in a carrier configuration). The catch-all in [trunkinbound] is just that: It catches all of them and turns control over to the agi script which uses the settings from the GUI.

Note that at NO time was this call in [default]. Anyone who suggests sending the call to [default] or that [default] is in any way related to inbound call management (outside the Vicidial Web GUI AFTER the inbound agi script has taken control) is possibly setting you up for an attack.

Here's how the attack works:

If someone can send a call to your server, they choose the DID it attempts to "dial" upon arrival. If this DID is dialed in [trunkinbound], the ONLY place it can go is to the inbound agi which is then managed by the vicidial GUI, which only allows the previously configured DIDs or the "default" DID which is preset to kill the call.

IF, however, this all takes place in the [default] context of extensions.conf instead of the [trunkinbound] context, a small miracle becomes possible: They could send the call to a non-existent DID which matches the pattern of an outbound carrier. This turns an inbound call from "whomever" into an outbound call paid for by ... YOU. Generally to the tone of several thousands of dollars per hour in international calls through a broker.

To review: The "Extension Context" in the DID configuration page is only used to route the call IF the DID in question is configured to go to an extension. This is very controlled, as it can not go to a random extension because control of the call has already passed to the DID agi script. Thus, the call will go only to the extension specified in the context specified. Nowhere else. In the preset configuration, this extension plays "not in service" and hangs up. It can't turn into an outbound call.

[edmund.uba]

I used sip setup debug on, and I have this when a call is coming through. Cant understand what it says.

[Aug 23 01:30:03] Reliably Transmitting (NAT) to 192.168.1.19:53718:
[Aug 23 01:30:03] OPTIONS sip:8009@192.168.1.19:53718;rinstance=c595631f0150c715;transport=UDP SIP/2.0
[Aug 23 01:30:03] Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK1c81a423;rport
[Aug 23 01:30:03] Max-Forwards: 70
[Aug 23 01:30:03] From: "asterisk" <sip:asterisk@192.168.1.3>;tag=as600de91a
[Aug 23 01:30:03] To: <sip:8009@192.168.1.19:53718;rinstance=c595631f0150c715;transport=UDP>
[Aug 23 01:30:03] Contact: <sip:asterisk@192.168.1.3:5060>
[Aug 23 01:30:03] Call-ID: 15686e376d000bce2b2b0a8303750fa7@192.168.1.3:5060
[Aug 23 01:30:03] CSeq: 102 OPTIONS
[Aug 23 01:30:03] User-Agent: Asterisk PBX 11.22.0-vici
[Aug 23 01:30:03] Date: Mon, 22 Aug 2016 17:30:03 GMT
[Aug 23 01:30:03] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug 23 01:30:03] Supported: replaces, timer
[Aug 23 01:30:03] Content-Length: 0
[Aug 23 01:30:03]
[Aug 23 01:30:03]
[Aug 23 01:30:03] ---
[Aug 23 01:30:03]
[Aug 23 01:30:03] <--- SIP read from UDP:192.168.1.19:53718 --->
[Aug 23 01:30:03] SIP/2.0 200 OK
[Aug 23 01:30:03] Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK1c81a423;rport=5060
[Aug 23 01:30:03] Contact: <sip:192.168.1.19:53718>
[Aug 23 01:30:03] To: <sip:8009@192.168.1.19:53718;rinstance=c595631f0150c715;transport=UDP>;tag=41409465
[Aug 23 01:30:03] From: "asterisk"<sip:asterisk@192.168.1.3>;tag=as600de91a
[Aug 23 01:30:03] Call-ID: 15686e376d000bce2b2b0a8303750fa7@192.168.1.3:5060
[Aug 23 01:30:03] CSeq: 102 OPTIONS
[Aug 23 01:30:03] Accept: application/sdp, application/sdp
[Aug 23 01:30:03] Accept-Language: en
[Aug 23 01:30:03] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
[Aug 23 01:30:03] Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
[Aug 23 01:30:03] User-Agent: Z 3.3.25608 r25552
[Aug 23 01:30:03] Allow-Events: presence, kpml
[Aug 23 01:30:03] Content-Length: 0
[Aug 23 01:30:03]
[Aug 23 01:30:03] <------------->
[Aug 23 01:30:03] --- (14 headers 0 lines) ---
[Aug 23 01:30:03] Really destroying SIP dialog '15686e376d000bce2b2b0a8303750fa7@192.168.1.3:5060' Method: OPTIONS
[Aug 23 01:30:04] Reliably Transmitting (NAT) to 69.71.197.20:5060:
[Aug 23 01:30:04] OPTIONS sip:69.71.197.20 SIP/2.0
[Aug 23 01:30:04] Via: SIP/2.0/UDP 180.191.87.171:5060;branch=z9hG4bK2cc962ae;rport
[Aug 23 01:30:04] Max-Forwards: 70
[Aug 23 01:30:04] From: "asterisk" <sip:asterisk@180.191.87.171>;tag=as6ab323d2
[Aug 23 01:30:04] To: <sip:69.71.197.20>
[Aug 23 01:30:04] Contact: <sip:asterisk@180.191.87.171:5060>
[Aug 23 01:30:04] Call-ID: 4a64468c0964f3820a4627552673581a@180.191.87.171:5060
[Aug 23 01:30:04] CSeq: 102 OPTIONS
[Aug 23 01:30:04] User-Agent: Asterisk PBX 11.22.0-vici
[Aug 23 01:30:04] Date: Mon, 22 Aug 2016 17:30:04 GMT
[Aug 23 01:30:04] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug 23 01:30:04] Supported: replaces, timer
[Aug 23 01:30:04] Content-Length: 0
[Aug 23 01:30:04]
[Aug 23 01:30:04]
[Aug 23 01:30:04] ---
[Aug 23 01:30:04] Reliably Transmitting (NAT) to 69.71.197.20:5060:
[Aug 23 01:30:04] OPTIONS sip:69.71.197.20 SIP/2.0
[Aug 23 01:30:04] Via: SIP/2.0/UDP 180.191.87.171:5060;branch=z9hG4bK1e3a964c;rport
[Aug 23 01:30:04] Max-Forwards: 70
[Aug 23 01:30:04] From: "asterisk" <sip:asterisk@180.191.87.171>;tag=as163b356e
[Aug 23 01:30:04] To: <sip:69.71.197.20>
[Aug 23 01:30:04] Contact: <sip:asterisk@180.191.87.171:5060>
[Aug 23 01:30:04] Call-ID: 6a51ddc5501479d304282b147e7f9203@180.191.87.171:5060
[Aug 23 01:30:04] CSeq: 102 OPTIONS
[Aug 23 01:30:04] User-Agent: Asterisk PBX 11.22.0-vici
[Aug 23 01:30:04] Date: Mon, 22 Aug 2016 17:30:04 GMT
[Aug 23 01:30:04] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug 23 01:30:04] Supported: replaces, timer
[Aug 23 01:30:04] Content-Length: 0
[Aug 23 01:30:04]
[Aug 23 01:30:04]
[Aug 23 01:30:04] ---
[Aug 23 01:30:04]
[Aug 23 01:30:04] <--- SIP read from UDP:69.71.197.20:5060 --->
[Aug 23 01:30:04] SIP/2.0 200 OK
[Aug 23 01:30:04] CSeq: 102 OPTIONS
[Aug 23 01:30:04] Via: SIP/2.0/UDP 180.191.87.171:5060;branch=z9hG4bK2cc962ae;rport
[Aug 23 01:30:04] From: "asterisk" <sip:asterisk@180.191.87.171>;tag=as6ab323d2
[Aug 23 01:30:04] Call-ID: 4a64468c0964f3820a4627552673581a@180.191.87.171:5060
[Aug 23 01:30:04] To: <sip:69.71.197.20>;tag=220830161702
[Aug 23 01:30:04] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
[Aug 23 01:30:04] Content-Length: 0
[Aug 23 01:30:04]
[Aug 23 01:30:04] <------------->
[Aug 23 01:30:04] --- (8 headers 0 lines) ---
[Aug 23 01:30:04] Really destroying SIP dialog '4a64468c0964f3820a4627552673581a@180.191.87.171:5060' Method: OPTIONS
[Aug 23 01:30:04]
[Aug 23 01:30:04] <--- SIP read from UDP:69.71.197.20:5060 --->
[Aug 23 01:30:04] SIP/2.0 200 OK
[Aug 23 01:30:04] CSeq: 102 OPTIONS
[Aug 23 01:30:04] Via: SIP/2.0/UDP 180.191.87.171:5060;branch=z9hG4bK1e3a964c;rport
[Aug 23 01:30:04] From: "asterisk" <sip:asterisk@180.191.87.171>;tag=as163b356e
[Aug 23 01:30:04] Call-ID: 6a51ddc5501479d304282b147e7f9203@180.191.87.171:5060
[Aug 23 01:30:04] To: <sip:69.71.197.20>;tag=220830161702
[Aug 23 01:30:04] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
[Aug 23 01:30:04] Content-Length: 0
[Aug 23 01:30:04]
[Aug 23 01:30:04] <------------->
[Aug 23 01:30:04] --- (8 headers 0 lines) ---
[Aug 23 01:30:04] Really destroying SIP dialog '6a51ddc5501479d304282b147e7f9203@180.191.87.171:5060' Method: OPTIONS
[Aug 23 01:30:06] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 23 01:30:06] == Manager 'sendcron' logged off from 127.0.0.1

[edmund.uba]

anyways, this is now my current carrier settings

register => XXXXXX:XXXXXXXXXX@69.71.197.20:5060/XXXXXX

[welco]
disallow=all
allow=ulaw
type=peer
username=XXXXXXXXXXXXXX
secret=XXXXXXXXXXXX
host=69.71.197.20
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=port,invite
nat=yes
externip=182.18.250.64

WTRUNK=SIP/welco

exten => _8.,1,Ringing
exten => _8.,2,Wait(1)
exten => _8.,3,Answer
exten => _8.,4,AGI(agi://127.0.0.1:4577/call_log)
exten => _8.,5,AGI(agi-DID_route.agi)
exten => _8.,6,Dial(${WTRUNK}/${EXTEN:1}@welco,55,o)
exten => _8.,7,Hangup

[edmund.uba]

this is the current DID setting

[mattyou1985]

ok onley thing i see difrent from mine is this

your, DID Route: IN_group
mine, is DID Route: AGENT

your, User Unavailable Action: voicemail
mine, User Unavailable Action: IN_gorup

your, User Route Settings In-Group: -none-
mine, User Route Settings In-Group: myingroupname

your, In-Group ID: youringroupname
mine, In-Group ID: -none-

your, No-Agent In-Group Redirect: DISABLED
mine, No-Agent In-Group Redirect: y

your, No-Agent In-Group ID: -none-
mine, No-Agent In-Group ID: myingroupname

now by dowing it this way a call will get rooted to the inbound que wether an agent is on or not so lease then you can see the call in the que

and hears a tip well a big one ask your pervider to ip lock your account as hackers can still sniff out passwords and usernames for your sip account this has happend to me and lots of others

[edmund.uba]

still not working, is my in group settings correct? Is it possible for the DID to be pointed at the ingroup instead than the agent? If so, what should be the correct settings?

Thanks.

how do I make the pictures bigger?

[mattyou1985]

em i carnt see sorry

as for fixing it i no its down to the size of the cut / paste frame
way i do mine is Prt Sc butten the open paint then Carl + V. then cut what you need Ctrl + C (to coppy agen) then new page then Ctrl + V (to paste it) and use https://postimage.org/ to upload it then it should be good

well yes its the same

look at this

DID Route: agent << needs

User Agent: << to work. if its blank it will then do next step

User Unavailable Action: IN_Group << so now it will look for the in group root

User Route Settings In-Group: YOUR_IN_GROUP_NAME << this is needed to tell it witch ingroup to go to

No-Agent In-Group Redirect: Y

No-Agent In-Group ID: YOUR_IN_GROUP_NAME

so now call arives PBX looks for an agent but we have not specified one to root to so the PBX will falow it next steps to root that call and the onley place we have told it to go if No-Agent and User Unavailable now point to YOUR_IN_GROUP_NAME the call will allways root to ingroup.

so now the call is out the DID root and now you should see the call beaing held in the ingoup you created.

why i did this well mine works ya so now lets say i change DID Route: to IN_Group

now for some reason when i call come in i hear the ring but the call dont get rooted to in_group never realey figerd out why i just found another way of doing it that works

hope this helps

and ime sorry to say ime not 100% using sip debug in the cli i normaley just look at the cli as thir is anoth info on thir to determin if the call is connecting to PBX

[williamconley]

...

exten => _8.,1,Ringing
exten => _8.,2,Wait(1)
exten => _8.,3,Answer
exten => _8.,4,AGI(agi://127.0.0.1:4577/call_log)
exten => _8.,5,AGI(agi-DID_route.agi)
exten => _8.,6,Dial(${WTRUNK}/${EXTEN:1}@welco,55,o)
exten => _8.,7,Hangup

Inbound does NOT require a "Dialplan Entry" ... this dialplan entry is not valid for outbound, and as such should be deleted.

I used sip setup debug on, and I have this when a call is coming through. Cant understand what it says.

[Aug 23 01:30:03] Reliably Transmitting (NAT) to 192.168.1.19:53718:
[Aug 23 01:30:03] OPTIONS sip:8009@192.168.1.19:53718;rinstance=c595631f0150c715;transport=UDP SIP/2.0
...
[Aug 23 01:30:06] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 23 01:30:06] == Manager 'sendcron' logged off from 127.0.0.1

This does not contain a call. Since there is no call during this debug, it's of no use at all. In most cases "sip debug" is not needed for troubleshooting inbound calls. Just the standard asterisk CLI output during a call is often enough. But your example shows NO output from the inbound call even with SIP debug on ... which means ...

In conclusion: Since you did turn on SIP Debug and did not get ANY activity during a call, the call is NOT getting to your Vicidial server. Either a firewall is in the way or the carrier is not sending the call to this server on a port asterisk is listening to. Ordinarily, asterisk listens on port 5060, so if the carrier is sending the calls to this server on port 5060, a firewall may be blocking your reception.