vicidial.org

Hi everyone,

I have created an Inbound campaign with a new carrier. I'm getting an issue, each time I'll dial the number to test the inbound campaign, the call automatically hang up but just before hanging up, I hear a voice message saying only "Inbound".

Can someone help me to sort this out plse. I have added the new carrier and configured it to do outbound and inbound.

Below are all my config:

VERSION: 2.14-708a
BUILD: 190414-0924
© 2019 ViciDial Group

SERVER + DESCRIPTION IP ACT LOAD CHAN AGNT DISK OUTBOUND INBOUND
vicibox-1 Server vicibox-1 0.0.0.0.0 Y / Y / Y 155 - 20% 68 18 9% LINK LINK

New Carrier Configuration:
Registration String: register=> 0230039805:xxxxxxxxxxx@0.0.0.0/xxxxxxxxxxxx

Account Entry:
type=friend
host=0.0.0.0
username=0230039805
secret=xxxxxxxxxxx
qualify=yes
fromdomain=0.0.0.0
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
nat=yes
context=trunkinbound
insecure=invite,port

Dial plan entry:
exten => _8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8X.,2,Dial(SIP/PLUGANDTEL/8*${EXTEN:2},,tTor)
exten => _8X.,3,Hangup

I have also created a New Ingroup / I have also added new DID's, created a new inbound campaign and associated 1 agent to them.

But unfortunately, it keep saying only Inbound each time I call the campaign DID.

I thank you in advance for your help and if you do need more info, feel free to ask.

Yours Sincerely
Rasbenny.

HI ,
you can post your Installation method to when you post so it will give some idea of others
In you sip carrier registered properly and your Incoming Dial plan and you incoming DID pointed to In group you tell your Settings details ?

first off, either take away nat=yes or change to nat=force_rport,comedia

second this dialplan seems off: exten => _8X.,2,Dial(SIP/PLUGANDTEL/8*${EXTEN:2},,tTor) - try change it to: exten => _8X.,2,Dial(SIP/PLUGANDTEL/${EXTEN:1},,tTor)

Hi,
Thank you very much for all you replies, I forgot to mention yesterday that I'm not a Vicidial genius but do uderstand the basics and some more stuffs.

@hunter2009: you can post your Installation method? Could you be more specific plse, I'm joining screenshots of all the settings I have made, hoping that it is the answer for your question. If not, don't hesitate to ask plse.

Plse find below the sreenshots of the In-group, DID and the inbound campaign settings:

https://1drv.ms/u/s!AufZBGOc8laVymrqvtkHrQB-Klt0?e=ROEdja

I hope that all these can help you to help me. Thanking you in advance for your co operation and your time.

Yours Sincerely
Rasbenny.

Hi,

@carpenox: I've done the changes you've asked, plse find below:

1. first off, either take away nat=yes or change to nat=force_rport,comedia

[PLUGANDTEL]
type=friend
host=
username=0230039805
secret=xxxxxxxxxxxxxxxx
qualify=yes
fromdomain=
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
nat=force_rport,comedia
context=trunkinbound
insecure=invite,port

2. second this dialplan seems off: exten => _8X.,2,Dial(SIP/PLUGANDTEL/8*${EXTEN:2},,tTor) - try change it to: exten => _8X.,2,Dial(SIP/PLUGANDTEL/${EXTEN:1},,tTor)
exten => _8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8X.,2,Dial(SIP/PLUGANDTEL/${EXTEN:1},,tTor)
exten => _8X.,3,Hangup

After which I have done a sip reload, a dialplan relod, a reload on the ssh.
Then, I have called 0230039805 using another platform (Avaya) but unfortunately there's still no ring tone when the number is dial and I still hear the voice message "Inbound" when the call reach my extension 39999.

Following which I have executed those command line on the ssh:
1. vicibox-1:~ # asterisk -rvvvv | grep "0230039805"
[Jun 9 11:27:58] -- Executing [0230039805@trunkinbound:1] AGI("SIP/PLUGANDTEL-00001740", "agi-DID_route.agi") in new stack

and then the command below to see the trace:

2. vicibox-1:~ # tcpdump -At -s0 -i eth1 host -w plug3.pcap
tcpdump: listening on eth1, link-type EN10MB (Ethernet), capture size 262144 bytes
^C126 packets captured
226 packets received by filter
0 packets dropped by kernel

As the forum don't allow more than 5 Url's, plse find the results on the following:

I hope that all these can help you to help me. Thanking you in advance for your co operation and your time.

Yours Sincerely
Rasbenny.

sounds like your carrier is not routing the calls to your server, do a "sip show peers" and make sure it shows as OK next to your carrier

also, remove that information, it has your Ip's and such that people can now use to attack your server and possibly hack into it, most likely if you do not have it secured properly

Hi,
Thank you again for your message.

PLse find below the result of the sip show peers command:

vicibox-1*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
30001/30001 (Unspecified) D Yes Yes 0 UNKNOWN
30002/30002 172.16.xx.xx D Yes Yes 50837 OK (255 ms)
30003/30003 (Unspecified) D Yes Yes 0 UNKNOWN
30004/30004 (Unspecified) D Yes Yes 0 UNKNOWN
30005/30005 172.16.xx.xx D Yes Yes 53774 OK (255 ms)
30006/30006 172.16.xx.xx D Yes Yes 55924 OK (361 ms)
30007/30007 (Unspecified) D Yes Yes 0 UNKNOWN
30008/30008 172.16.xx.xx D Yes Yes 50267 OK (255 ms)
30009/30009 172.16.xx.xx D Yes Yes 51609 OK (255 ms)
30010/30010 (Unspecified) D Yes Yes 0 UNKNOWN
30012/30012 172.16.xx.xx D Yes Yes 59771 OK (341 ms)
30013/30013 (Unspecified) D Yes Yes 0 UNKNOWN
30014/30014 (Unspecified) D Yes Yes 0 UNKNOWN
30015/30015 (Unspecified) D Yes Yes 0 UNKNOWN
30016/30016 (Unspecified) D Yes Yes 0 UNKNOWN
30017/30017 (Unspecified) D Yes Yes 0 UNKNOWN
30018/30018 (Unspecified) D Yes Yes 0 UNKNOWN
30019/30019 172.16.xx.xx D Yes Yes 51123 OK (1255 ms)
30020/30020 172.16.xx.xx D Yes Yes 61883 OK (255 ms)
30021/30021 172.16.xx.xx D Yes Yes 63615 OK (255 ms)
30022/30022 172.16.xx.xx D Yes Yes 50277 OK (255 ms)
30023/30023 (Unspecified) D Yes Yes 0 UNKNOWN
30024/30024 (Unspecified) D Yes Yes 0 UNKNOWN
30025/30025 (Unspecified) D Yes Yes 0 UNKNOWN
30026/30026 172.16.xx.xx D Yes Yes 55870 OK (255 ms)
30027/30027 (Unspecified) D Yes Yes 0 UNKNOWN
30028/30028 172.16.xx.xx D Yes Yes 51707 OK (255 ms)
30029/30029 172.16.xx.xx D Yes Yes 58903 OK (271 ms)
30030/30030 (Unspecified) D Yes Yes 0 UNKNOWN
30098/30098 172.16.xx.xx D Yes Yes 61765 OK (329 ms)
30099/30099 (Unspecified) D Yes Yes 0 UNKNOWN
39999/39999 172.16.xx.xx D Yes Yes 55924 OK (255 ms)
40001/40001 10.0.xx.xx D Yes Yes 62574 OK (38 ms)
40002/40002 (Unspecified) D Yes Yes 0 UNKNOWN
40003/40003 (Unspecified) D Yes Yes 0 UNKNOWN
40004/40004 (Unspecified) D Yes Yes 0 UNKNOWN
40005/40005 (Unspecified) D Yes Yes 0 UNKNOWN
40006/40006 (Unspecified) D Yes Yes 0 UNKNOWN
40007/40007 (Unspecified) D Yes Yes 0 UNKNOWN
40008/40008 10.0.xx.xx D Yes Yes 61180 OK (37 ms)
40009/40009 (Unspecified) D Yes Yes 0 UNKNOWN
40010/40010 10.0.xx.xx D Yes Yes 55619 OK (40 ms)
60001/60001 (Unspecified) D Yes Yes 0 UNKNOWN
PLUGANDTEL/0230039805 0.0.0.0 Yes Yes 5060 OK (10 ms)
activsol/activsol 0.0.0.0 Yes Yes A 5060 OK (5 ms)
45 sip peers [Monitored: 20 online, 25 offline Unmonitored: 0 online, 0 offline]
[Jun 9 16:09:11] == Manager 'sendcron' logged on from 127.0.0.1
[Jun 9 16:09:11] -- Called 58600052@default
[Jun 9 16:09:11] -- Executing [58600052@default:1] MeetMe("Local/58600052@default-00000877;2", "8600052,Fmq") in new stack
[Jun 9 16:09:11] -- Local/58600052@default-00000877;1 answered
[Jun 9 16:09:11] -- Executing [8309@default:1] Answer("Local/58600052@default-00000877;1", "") in new stack
[Jun 9 16:09:11] -- Executing [8309@default:2] Monitor("Local/58600052@default-00000877;1", "wav,5086_20210609-160909_30005_0610835531") in new stack
[Jun 9 16:09:11] -- Executing [8309@default:3] Wait("Local/58600052@default-00000877;1", "3600") in new stack
[Jun 9 16:09:11] -- <Local/910610835531@default-00000872;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[Jun 9 16:09:12] == Manager 'sendcron' logged off from 127.0.0.1
vicibox-1*CLI>
Disconnected from Asterisk server
[Jun 9 16:09:12] Asterisk cleanly ending (0).
[Jun 9 16:09:12] Executing last minute cleanups

As you can see, it says OK. What test or trace can I do to see what's going wrong plse?

Yours Sincerely
Rasbenny

i dont see you typing "Exit", is your asterisk crashing?

@rasbenny

as per your logs the call is landing in the server and ingroup .
1. vicibox-1:~ # asterisk -rvvvv | grep "0230039805"
[Jun 9 11:27:58] -- Executing [0230039805@trunkinbound:1] AGI("SIP/PLUGANDTEL-00001740", "agi-DID_route.agi") in new stack

the voice inbound is playing because you have set Agent Alert Filename: inbound.

can you post full cli log of 0230039805@trunkinbound:
ie log of the call till it reaches or rings the agent phone.

are you using ON-HOOK enabled in agents phones?

Hi,
@carpenox, No my saterisk does not crash, I do either Ctrl+c or I type Exit.

Thanks again for your kind help and reply

Yours Sincerely
Rasbenny

Hi,
@ Striker, thanks for your reply and your time,

For the extension 39999, on the page MODIFY A PHONE RECORD: 39999, The option On-Hook Agent is set to N.

full cli log of 0230039805@trunkinbound:
Jai celui la, jespere que cela vas vous aider, sinon j'aurai besoin que vous me donniez la ligne de commande pour sortir le full log 0230039805@trunkinbound:

Jau utiliser : vicibox-1*CLI> cat full | grep "0230039805"
No such command 'cat full | grep "0230039805"' (type 'core show help cat full' for other possible commands)

Je vous mets le resultats que j'ai entre temps:
https://1drv.ms/t/s!AufZBGOc8laVywW1ikoqWydwQKCb?e=TJYZQi

Thanks again for everything you are doing.

Yours Sincerely
Rasbenny

Hi Rasbenny

1. while dialing in , did you able to listen to the voice played by "welcome_fr.gsm" which is welcome message you have set.

2. "inbound" voice is played to callee or the 39999 extension (agent)?

Just a try to check call without ingroup
point your DID to Phone: DID route : Phone
and in phone field set 39999

Dial in and check calls between 39999 and inbound DID calll,
post back your outputs here.

also noticed
0*0*0*0 ip in log, is it something you edited before posting here?
few phones shows 10.0 and 172.16 network
can you post more detail about your network setup

Hi,
@Striker,
1. while dialing in , did you able to listen to the voice played by "welcome_fr.gsm" which is welcome message you have set. - No I dont hear the welcome message, only the word Inbound.

2. "inbound" voice is played to callee or the 39999 extension (agent)? The callee is myself as well as the 39999 extension agent. I'm the only one to be logged onto that inbound campaign. I make somebody else who is using Avaya on another server or I take a hard phone and make someone dial the DID 0230039805. When I receive the call, I hear only the word "Inbound", nothing else. The counter keeps turning as if the call is still going on but nor the caller nor me can hear each other. On the caller's side when he dialed the number he didn't hear any dial tone neither.

3. One friend @hunter2009 configuered eyebeam directly with the adress of the new carrier, without going through vicidial. when I make someone dial the DID it worked perfectly. There were a dial tone, a ring tone and we were able to talk to each other on the line.

4. So now I'm sure that the issue is with the configuration of the DID on vicidial but the thing is that I do have another DID with another carrier which is working well, I've used it as a model to configure the new one, for the ingroup, i've used the same old ingroup which worked with the old carrier. Even for the campaign, I've tested everything with the old campaign on which I still receives calls. But each time I get the same issue when calling the DID.

5. DID route : Phone and in phone field set 39999:

https://1drv.ms/t/s!AufZBGOc8laVyzNiCJTROvfZ9KEd?e=jEW1gT

6. also noticed 0*0*0*0 ip in log, is it something you edited before posting here? : Yes it was me who edited it, i've remove all ip's. But if you need those info to investigate, feel free to ask plse.

7. few phones shows 10.0 and 172.16 network can you post more detail about your network setup: This is because the phones with ip adresses 10.0... are in Tunis and those with 172.16... are in Mauritius.

I must admit that from those past two days I have learn more on vicidial than all that I've been through this year. I'm very happy about it but I would have been more happier if I can finally sor this out.

I thank you again for all your help and your time.

Yours Sincerely
Rasbenny.

seems lack of rtp packets, Nat issue.

if possible pm your network setup, with IP of server and IP of the carrier or post here with masked ip details.