2 nics , internet sharing and remote access help

[silverstone]

hi ppl ,
i have a new vicidialnow 1.3 ce installation done
1. i have 2 nics installed
i want one nic fo WAN and 1 for my LAN
what i have done is
eth0
ipaddr=172.24.***.***
netmask=255.255.255.0
gateway=172.24.2.1

eth1
ipaddr=192.168.1.100
netmask=255.255.255.0
gateway=172.24.***.***
now my question is i want internet also in my client sytems .. the ip in one of my client system is 192.168.1.101 and gateway is 192.168.1.100
and DNS i have given my isp dns address
but stilll internet s not working
i require help on this ..
2> i cant ping the WAN ip from anywhere else
why is that ..
anything to be done in sip.conf or extensions .conf if yes then pls mention what changes
any help at the earliest possible would be appreciated
thank u guys

[gmcust3]

Try this :

https://www.centos.org/modules/newbb/vi ... 5&forum=40

[djprice]

Hi Guys,

New poster so please bear with me but here goes:

Using VICIDial Now updated to 2.2.1-237
asterisk 1.2
Codec G729
Server P4 Dual Core, 2GB RAM, 500GB HDD
UK SIP Provider (Voiceflex)
CentOS
Load Avg - not that far yet.

OK, the server has 2NICS and is setup in a similar way to several trixbox installations I have running. What I mean by this is one NIC is on a 192.168.1.0 network which all the client PCs are on, the other is 192.168.11.0 network connected only to a broadband router (Draytek Vigor) The default gateway of the machine is set to 192.168.11.1. The 192.168.1.1 network also has an internet connection which is used for non-voice data. I know there are probably better ways of segregating the Voice IP traffic and I am open to suggestions. However as it is I can dial out directly from the the softphones (xlite) and I can manually dial through VICI dial both through Dial next and the manual dial link.

However when I try to use ratio or adapt dialing it shows the agents logged in and waiting for calls, but does not actually place any calls.

I did initially try with just one NIC and that worked. When I enabled the second as described above it makes no attempt to dial.

Is this likely to be caused by the setup with the default gateway? If so I would thoroughly appreciate being pointed in the direction of how I should do this.

Also when I run update-server-ip if I set the current to 192.168.11.2 (on what I want as the gateway for the voip traffic) the soft phones do not ring when an agent logs in. If I set it to 192.168.1.2 that part works correctly.

Thanks in advance for any advice and your patience.

[djprice]

Should probably have included the dialplan:
exten => _9XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:1},,tToR)
exten => _9XXXXXXXXXXX,3,Hangup


Asterisk Output:

Connected to Asterisk 1.2.30.2 currently running on vici (pid = 2553)
[May 12 22:41:14] -- Remote UNIX connection
Verbosity is at least 21
[May 12 22:41:50] -- Registered SIP 'cc100' at 192.168.1.57 port 50680 expir es 3600
[May 12 22:41:50] -- Saved useragent "X-Lite release 1002tx stamp 29712" for peer cc100
[May 12 22:42:01] == Parsing '/etc/asterisk/manager.conf': [May 12 22:42:01] Found
[May 12 22:42:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 22:42:01] == Parsing '/etc/asterisk/manager.conf': [May 12 22:42:01] Found
[May 12 22:42:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 22:42:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 22:42:06] == Parsing '/etc/asterisk/manager.conf': [May 12 22:42:06] Found
[May 12 22:42:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 22:42:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 22:42:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 22:43:01] == Parsing '/etc/asterisk/manager.conf': [May 12 22:43:01] Found
[May 12 22:43:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 22:43:01] == Parsing '/etc/asterisk/manager.conf': [May 12 22:43:01] Found
[May 12 22:43:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 22:43:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 22:43:06] == Parsing '/etc/asterisk/manager.conf': [May 12 22:43:06] Found
[May 12 22:43:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 22:43:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 22:43:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 22:43:07] == Parsing '/etc/asterisk/manager.conf': [May 12 22:43:07] Found
[May 12 22:43:07] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 22:43:10] > Channel SIP/cc100-09fa0330 was answered.
[May 12 22:43:10] -- Executing MeetMe("SIP/cc100-09fa0330", "8600052|F") in new stack
[May 12 22:43:10] == Parsing '/etc/asterisk/meetme.conf': [May 12 22:43:10] Found
[May 12 22:43:10] == Parsing '/etc/asterisk/meetme-vicidial.conf': [May 12 22:43:10] Found
[May 12 22:43:10] -- Created MeetMe conference 1023 for conference '8600052'
[May 12 22:43:10] -- Playing 'conf-onlyperson' (language 'en')
[May 12 22:43:11] == Manager 'sendcron' logged off from 127.0.0.1
vici*CLI>



And campaign screen
DIAL LEVEL: 10 TRUNK SHORT/FILL: / 0 FILTER: NONE TIME: 2010-05-12 22:44:48
DIALABLE LEADS: 29690 CALLS TODAY: 0 AVG AGENTS: 0 DIAL METHOD: ADAPT_TAPERED
HOPPER LEVEL: 100 DROPPED / ANSWERED: 0 / 0 DL DIFF: 0 STATUSES: NEW
LEADS IN HOPPER: 100 DROPPED PERCENT: 0% DIFF: 0.00% ORDER: DOWN
+ VIEW MORE VIEW USER GROUP SHOW SERVER INFO HIDE WAITING CALLS SHOW IN-GROUP STATS SHOW PHONES SHOW CUSTPHONES
NO LIVE CALLS WAITING
1 agents logged in 0 agents in calls 1 agents waiting 0 paused agents 0 agents in dead calls

VICIDIAL: Agents Time On Calls Campaign: |TESTCAMP| 2010-05-12 22:44:48
+----------------+------------------------+-----------+----------+---------+------------+-------+------+------------------
| STATION | USER SHOW ID INFO | SESSIONID | STATUS | MM:SS | CAMPAIGN | CALLS | HOLD | IN-GROUP
+----------------+------------------------+-----------+----------+---------+------------+-------+------+------------------
| SIP/cc100 | Admin + | 8600052 | READY | 21228344:48 | TESTCAMP | 0 |
+----------------+------------------------+-----------+----------+---------+------------+-------+------+------------------
1 agents logged in on all servers
System Load Average: 0.22 0.10 0.02

- Agent waiting for call
- Agent waiting for call > 1 minute
- Agent waiting for call > 5 minutes
- Agent on call > 10 seconds
- Agent on call > 1 minute
- Agent on call > 5 minutes
- Agent Paused > 10 seconds
- Agent Paused > 1 minute
- Agent Paused > 5 minutes
- Agent in 3-WAY > 10 seconds
- Agent on a dead call


Hope this isn't information overload.

Thanks again

[williamconley]

Try this :

https://www.centos.org/modules/newbb/vi ... 5&forum=40

did you try this?

[djprice]

Seem to be having trouble posting

[djprice]

Will try posting in stages

I had read that but my understanding is it is trying to solve a different problem.

I believe they are trying to use a CentOs box as a router to share their internet connection.

In the example I gave I was trying to use the default gateway of the second NIC in the Vicidialnow box force internet traffic (i.e. VoIP traffic) for that particular machine out of one LAN and receive internal network traffic (from agents' pcs on the same LAN), where there is another internet connection for general browsing etc.

[djprice]

This is a setup I have successfully working with Trixbox (also based on CentOS) and as far as I can see, traffic wise it is working here as I can make calls through the Draytek interface and they connect fine.

What I am really asking is, "is there anything in this setup which will break the Vicidial ratio or adapt dial methods?" because, as I said, it knows agents are available, it just does not dial anything.

[djprice]

For ease of understanding I have drawn up a quick network topography diagram which shows how it is all wired, but the forum won't let me post a link as I am a new member. I can email this or pm a link.

As I say I am looking for a sanity check on the method to know whether I am wasting my time trying to get it working.

Thanks again.

[djprice]

well after all that it was because I used the word " U R L " in the post, thats why it was blocking me

[williamconley]

vicidial dials through asterisk. if it works in asterisk, it works.

vicidial DOES care what the local ip address is, if that changes you must run the admin server ip address change script (in /usr/share/astguiclient)

[djprice]

Hi

The machine has 2 local IPs

How would this affect it?

Like I say I can make calls etc, it just does not autodial.

I am assuming it is because of the two NICs because I had it working on just one NIC.

What else should I look out for? Already checked 24 hour dialling and the obvious stuff from the forums. Agents are hitting Resume etc.

I have looked in MySQL and the vicidial_auto_calls table is empty.

Is there anything less obvious I may be missing?

Thanks

[williamconley]

standard troubleshooting ...

check to see if it is attempting to dial

astguiclient logs

asterisk CLI/asterisk logs

[djprice]

Asterisk log below, more to follow

Connected to Asterisk 1.2.30.2 currently running on vici (pid = 2553)
[May 13 16:54:01] -- Remote UNIX connection
Verbosity is at least 21
[May 13 16:54:01] == Parsing '/etc/asterisk/manager.conf': [May 13 16:54:01] Found
[May 13 16:54:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 16:54:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 16:54:06] == Parsing '/etc/asterisk/manager.conf': [May 13 16:54:06] Found
[May 13 16:54:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 16:54:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 16:54:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 16:54:14] == Parsing '/etc/asterisk/manager.conf': [May 13 16:54:14] Found
[May 13 16:54:14] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 16:54:14] == Parsing '/etc/asterisk/manager.conf': [May 13 16:54:14] Found
[May 13 16:54:14] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 16:54:14] -- Executing MeetMeAdmin("Local/55558600051@default-be3e,2", "8600051|K") in new stack
[May 13 16:54:14] -- Hungup 'Zap/pseudo-1636925066'
[May 13 16:54:14] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-09f9ba98'
[May 13 16:54:14] -- Executing DeadAGI("SIP/cc100-09f9ba98", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[May 13 16:54:14] == Parsing '/etc/asterisk/meetme.conf': [May 13 16:54:14] Found
[May 13 16:54:14] == Parsing '/etc/asterisk/meetme-vicidial.conf': [May 13 16:54:14] Found
[May 13 16:54:14] NOTICE[27675]: app_meetme.c:2210 admin_exec: Conference Number not found
[May 13 16:54:14] -- Executing Hangup("Local/55558600051@default-be3e,2", "") in new stack
[May 13 16:54:14] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-be3e,2'
[May 13 16:54:14] -- Executing DeadAGI("Local/55558600051@default-be3e,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[May 13 16:54:14] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 13 16:54:14] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 13 16:54:18] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 16:54:18] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 16:54:27] == Parsing '/etc/asterisk/manager.conf': [May 13 16:54:27] Found
[May 13 16:54:27] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 16:54:32] > Channel SIP/cc100-09fa0330 was answered.
[May 13 16:54:32] -- Executing MeetMe("SIP/cc100-09fa0330", "8600051|F") in new stack
[May 13 16:54:32] == Parsing '/etc/asterisk/meetme.conf': [May 13 16:54:32] Found
[May 13 16:54:32] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 16:54:32] == Parsing '/etc/asterisk/meetme-vicidial.conf': [May 13 16:54:32] Found
[May 13 16:54:32] -- Created MeetMe conference 1023 for conference '8600051'
[May 13 16:54:32] -- Playing 'conf-onlyperson' (language 'en')
vici*CLI>

[djprice]

Now when I have loaded the campaign realtime it is showing:

DIAL LEVEL: 10 TRUNK SHORT/FILL: / 0 FILTER: NONE TIME: 2010-05-13 18:08:55
DIALABLE LEADS: 0 CALLS TODAY: 0 AVG AGENTS: 0 DIAL METHOD: ADAPT_TAPERED
HOPPER LEVEL: 100 DROPPED / ANSWERED: 0 / 0 DL DIFF: 0 STATUSES: NEW
LEADS IN HOPPER: 100 DROPPED PERCENT: 0% DIFF: 0.00% ORDER: DOWN

However there are 50,000 leads in the list which have not been dialed. This has changed since last night.

in the lead table I have 50K rows of:

Full Texts lead_id entry_date modify_date status user vendor_lead_code source_id list_id gmt_offset_now called_since_last_reset phone_code phone_number title first_name middle_initial last_name address1 address2 address3 city state province postal_code country_code gender date_of_birth alt_phone email security_phrase comments called_count last_local_call_time rank owner
Edit Delete 1 2010-05-12 20:17:18 0000-00-00 00:00:00 NEW 101 1.00 N 44 07968000000 0000-00-00 0 2008-01-01 00:00:00 0
Edit Delete 2 2010-05-12 20:17:18 0000-00-00 00:00:00 NEW 101 1.00 N 44 07968000001 0000-00-00 0 2008-01-01 00:00:00 0
Edit Delete 3 2010-05-12 20:17:18 0000-00-00 00:00:00 NEW 101 1.00 N 44 07968000002 0000-00-00 0 2008-01-01 00:00:00 0
Edit Delete 4 2010-05-12 20:17:18 0000-00-00 00:00:00 NEW 101 1.00 N 44 07968000003 0000-00-00 0 2008-01-01 00:00:00 0
Edit Delete 5 2010-05-12 20:17:18 0000-00-00 00:00:00 NEW 101 1.00 N 44 07968000004 0000-00-00 0 2008-01-01 00:00:00 0
Edit Delete 6 2010-05-12 20:17:18 0000-00-00 00:00:00 NEW 101 1.00 N 44 07968000005 0000-00-00 0 2008-01-01 00:00:00 0


I may be wrong but I am reading that as not dialled?

Any suggestions would be greatly appreciated.

Thanks

[djprice]

How does the fact that there are 2 IPs affect vicidial. Do I need to change anything specifially for this?

At the moment update_server_ip is set to 192.168.1.2 (the local network to the workstations and soft phones)

I have tried it at 192.168.11.2 to match the NIC connected to the Draytek router but then the softphones did not ring when I logged an agent in.

[williamconley]

Code: Select all

ip route show

[djprice]

[root@vici ~]# ip route show
192.168.1.0/24 dev eth0 proto kernel scope link src 192.168.1.2
192.168.11.0/24 dev eth1 proto kernel scope link src 192.168.11.2
default via 192.168.11.1 dev eth1

[williamconley]

Code: Select all

ping xxx.xxx.xxx.xxx -c 1

(replace x's with your VOIP provider)

[djprice]

[root@vici ~]# ping 146.101.248.200 -c 1
PING 146.101.248.200 (146.101.248.200) 56(84) bytes of data.
64 bytes from 146.101.248.200: icmp_seq=1 ttl=56 time=22.1 ms

--- 146.101.248.200 ping statistics ---
1 packets transmitted, 1 received, 0% packet loss, time 0ms
rtt min/avg/max/mdev = 22.166/22.166/22.166/0.000 ms

[williamconley]

ok, so let's log in an agent and go through it one step at a time and find out what happens.

cli from an agent logging in ... including confirmatio that the agent hears "you are the only person in this conference".

[djprice]

Verbosity is at least 21
[May 13 20:05:19] == Parsing '/etc/asterisk/manager.conf': [May 13 20:05:19] Found
[May 13 20:05:19] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:05:22] > Channel SIP/cc100-09f9ba98 was answered.
[May 13 20:05:22] -- Executing MeetMe("SIP/cc100-09f9ba98", "8600051|F") in new stack
[May 13 20:05:22] == Parsing '/etc/asterisk/meetme.conf': [May 13 20:05:22] Found
[May 13 20:05:22] == Parsing '/etc/asterisk/meetme-vicidial.conf': [May 13 20:05:22] Found
[May 13 20:05:22] -- Created MeetMe conference 1023 for conference '8600051'
[May 13 20:05:22] -- Playing 'conf-onlyperson' (language 'en')
[May 13 20:05:23] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 20:06:01] == Parsing '/etc/asterisk/manager.conf': [May 13 20:06:01] Found
[May 13 20:06:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:06:01] == Parsing '/etc/asterisk/manager.conf': [May 13 20:06:01] Found
[May 13 20:06:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:06:01] == Manager 'sendcron' logged off from 127.0.0.1





Agent hears "only person inconference

[williamconley]

attempt a manual dial from within vici agent screen and show cli

[djprice]

Verbosity is at least 21
[May 13 20:12:57] -- Remote UNIX connection
[May 13 20:13:01] == Parsing '/etc/asterisk/manager.conf': [May 13 20:13:01] Found
[May 13 20:13:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:13:01] == Parsing '/etc/asterisk/manager.conf': [May 13 20:13:01] Found
[May 13 20:13:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:13:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 20:13:03] == Parsing '/etc/asterisk/manager.conf': [May 13 20:13:03] Found
[May 13 20:13:03] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:13:03] -- Executing MeetMe("Local/8600051@default-91ed,2", "8600051|F") in new stack
[May 13 20:13:03] > Channel Local/8600051@default-91ed,1 was answered.
[May 13 20:13:03] -- Executing AGI("Local/8600051@default-91ed,1", "agi://127.0.0.1:4577/call_log") in new stack
[May 13 20:13:03] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 13 20:13:03] -- Executing Dial("Local/8600051@default-91ed,1", "SIP/84415841/07748171925||tToR") in new stack
[May 13 20:13:03] -- Called 84415841/0774817xxxx
[May 13 20:13:07] == Parsing '/etc/asterisk/manager.conf': [May 13 20:13:07] Found
[May 13 20:13:07] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:13:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 20:13:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 20:13:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 20:13:14] -- SIP/84415841-09fa90f8 is making progress passing it to Local/8600051@default-91ed,1
[May 13 20:13:17] -- SIP/84415841-09fa90f8 answered Local/8600051@default-91ed,1


Call gets through fine and can hear on both ends


(thanks for the help btw thoroughly appreciated)

[williamconley]

verify leads in the hopper for this campaign

verify leads available to be dialed for this campaign

[djprice]

100 leads in the hopper

However dialable leads are showing as 0 (they weren't last night when I first posted)

Records in the leads table all look like:

Edit Delete 1 2010-05-12 20:17:18 0000-00-00 00:00:00 NEW 101 1.00 N 44 07968000000 0000-00-00 0 2008-01-01 00:00:00 0


Records in hopper look like:

1 1 TESTCAMP READY 101 1.00 NONE 0


What does priority 0 mean?
Is this affecting anything?

Date seems to be correct on the server Thu May 13 20:28:44 BST 2010

[williamconley]

stick with me now ... if you have NO leads to be dialed, your server will not dial.

you must get leads available to be dialed.

do you know why there are none?

[djprice]

There were leads there earlier. Is it dummy dialing them? as in going through the motions and not dialing?

I had the same prob with 27000 leads available last night (see posts above) but is still wasn't dialing.

The only thing I can think of it is doing part of the process through the scripts but not passing the calls to Asterisk.

How do I fix this?

Thanks

[djprice]

Just uploaded my number 80 times on top of the 50K. Still showing no leads to dial on the realtime screen

but campaign screen shows:

his campaign has 1 active lists and 0 inactive lists

|| This campaign has 50081 leads to be dialed in those lists - HIDE

This campaign has 100 leads in the dial hopper


Logged in again and still no activity:

Connected to Asterisk 1.2.30.2 currently running on vici (pid = 2553)
[May 13 20:58:38] -- Remote UNIX connection
Verbosity is at least 21
[May 13 20:58:51] == Parsing '/etc/asterisk/manager.conf': [May 13 20:58:51] Found
[May 13 20:58:51] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:58:52] == Parsing '/etc/asterisk/manager.conf': [May 13 20:58:52] Found
[May 13 20:58:52] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:58:52] -- Executing MeetMeAdmin("Local/55558600051@default-c1eb,2", "8600051|K") in new stack
[May 13 20:58:52] == Parsing '/etc/asterisk/meetme.conf': [May 13 20:58:52] Found
[May 13 20:58:52] == Parsing '/etc/asterisk/meetme-vicidial.conf': [May 13 20:58:52] Found
[May 13 20:58:52] NOTICE[28199]: app_meetme.c:2210 admin_exec: Conference Number not found
[May 13 20:58:52] -- Executing Hangup("Local/55558600051@default-c1eb,2", "") in new stack
[May 13 20:58:52] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-c1eb,2'
[May 13 20:58:52] -- Executing DeadAGI("Local/55558600051@default-c1eb,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[May 13 20:58:52] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 13 20:58:55] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 20:58:56] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 20:58:58] == Parsing '/etc/asterisk/manager.conf': [May 13 20:58:58] Found
[May 13 20:58:58] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:59:00] > Channel SIP/cc100-09f9ba98 was answered.
[May 13 20:59:00] -- Executing MeetMe("SIP/cc100-09f9ba98", "8600051|F") in new stack
[May 13 20:59:00] == Parsing '/etc/asterisk/meetme.conf': [May 13 20:59:00] Found
[May 13 20:59:00] == Parsing '/etc/asterisk/meetme-vicidial.conf': [May 13 20:59:00] Found
[May 13 20:59:00] -- Created MeetMe conference 1023 for conference '8600051'
[May 13 20:59:00] -- Playing 'conf-onlyperson' (language 'en')
[May 13 20:59:01] == Parsing '/etc/asterisk/manager.conf': [May 13 20:59:01] Found
[May 13 20:59:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:59:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 20:59:01] == Parsing '/etc/asterisk/manager.conf': [May 13 20:59:01] Found
[May 13 20:59:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:59:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 20:59:06] == Parsing '/etc/asterisk/manager.conf': [May 13 20:59:06] Found
[May 13 20:59:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 20:59:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 20:59:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 21:00:01] == Parsing '/etc/asterisk/manager.conf': [May 13 21:00:01] Found
[May 13 21:00:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 21:00:01] == Parsing '/etc/asterisk/manager.conf': [May 13 21:00:01] Found
[May 13 21:00:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 21:00:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 21:00:06] == Parsing '/etc/asterisk/manager.conf': [May 13 21:00:06] Found
[May 13 21:00:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 21:00:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 21:00:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 13 21:00:10] == Refreshing DNS lookups.

[williamconley]

the real time screen wont catch up for a while, not interested in that. the campaign screen showing 50081 is where it counts.

so the agent logged in to the campaign, then the agent pressed "resume"?

[djprice]

Yes, didn't fall for that one, they are hitting resume.

I did have a test setup running with 3 agents for 20 mins at the end of a shift 2 days ago.

Before I introduced the second network card.

[williamconley]

ok, now you need to verify the dial prefix in the campaign is valid. verify the dial code and phone numbers in the list are loading valid (as in: dial from the asterisk command line or a phone on the system ... [prefix][code][phone_number]) to verify that the number it is "theoretically" attempting to dial should result in a phone call.

[djprice]

When I set the campaign to manual it works:

My Dial Plan is:
exten => _9XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:1},,tToR)
exten => _9XXXXXXXXXXX,3,Hangup

I have
Dial Prefix: 9
Omit Phone Code: yes

in the campaign

If it dials out in manual mode does that mean it should work in ratio?

[williamconley]

no

[williamconley]

what are your dial prefix, dial code, and phone number? (xxxx the last 4 if you like)

for BOTH manual dial and auto dial

also, when you dialed manual, did you enter the number manually or "dial next lead"?

[djprice]

Prefix 9 code 44 (set to ignore in campaign) number 07748171XXX

So to dial through manual dial I type 907748171xxx

clicking on dial next in manual mode also works

Manual dial next to a "It has not been possible to connect your call" telco message below:

[May 13 21:55:08] == Manager 'sendcron' logged on from 127.0.0.1
[May 13 21:55:08] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-1d28,2'
[May 13 21:55:08] -- Executing DeadAGI("Local/8600051@default-1d28,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 13 21:55:08] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 13 21:55:08] == Spawn extension (default, 907968000001, 2) exited non-zero on 'Local/8600051@default-1d28,1'
[May 13 21:55:08] -- Executing DeadAGI("Local/8600051@default-1d28,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
[May 13 21:55:08] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 13 21:55:10] == Refreshing DNS lookups.
[May 13 21:55:12] == Manager 'sendcron' logged off from 127.0.0.1

[djprice]

Don't know if it helps but after those few manual dials I now have:

DIALABLE LEADS: 50080

on the real time screen.

Still not ratio dialling though

[williamconley]

just to be clear: "clicking on dial next in manual mode also works"

but you get "It has not been possible to connect your call" which you consider to be "working" because you heard that expected recording? (Not because a call was connected?) right?

(Please, in the future, use a test list with YOUR CELL, because when we finish here ... the next thing could easily be "one way sound" and then you have to reload the list, LOL). Control the circumstances of any test scenario by controlling everything you can, including the answering of the phone, your ability to interact with the in as many ways as possible.

Moving right along ...

I feel the urgent need to clarify something for you weirdo UK people (I've bumped into this a few times, and it never ceases to catch me off guard!):

your "dial code" (if dialing the UK from the UK) is NOT 44. It is "0". Your phone numbers are 10 digit phone numbers. They are preceded by the "dial code" of "0". Here in the US, we have the SAME SETUP, but our "dial code" is "1".

So: HERE: 13214268368 THERE 03214268368, same phone number but different dial code because we have "1" and you have "0" for dialing inside the country.

If you were planning to dial a number in the US or some other country, you would dial a completely different dial code before the number which would include the country code and any other required digits to get to that phone number. For instance if I were dialing that same number

So please investigate that in your list, load one with the appropriate 10 digit phone numbers, dial code populated correctly with a "0", and set your dial prefix to "9" if that is the carrier you have set up.

Then we can jump into whether your dial plan is messed up or whether your dialer is not actually trying to dial at all for some reason.

this would be accomplished via "show dialplan 90XXXXXXXXXX" replacing the Xs with a real number

then it is time to verify the settings in the campaign allow for dialing and check the astguiclient logs for "available agents" and it's decision making process for whether or not to dial. I also recommend (at this point) creating a new campaign with only those 80 fresh cell phone leads on it in a fresh list (fresh list, fresh campaign, fresh leads) on the off chance that there was damage to the campaig record.

[djprice]

Hi Mate


The not possible to connect is a uk mobile message so the call is getting out to the mobile network.

I think the confusion for us UK weirdos is we are thinking dial code as international dial code (uk is 44, so you would dial 0044 to dial a uk number or from a mobile +44) We know that the US is +1 which is why we get confused.

Applied the changes you suggested, dropped the leading zeros from the sample file, chose omit phone code NO and the current dial plan is

exten => _90XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _90XXXXXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:1},,tToR)
exten => _90XXXXXXXXXX,3,Hangup


Same result, can manually dial myself clicking dial next and hold a 2 way conversation with myself.

When hitting ratio or adapt shows 80 nos available, 50 leads in the hopper, 1 agent available and no current dialling activity.

Any help would be greatly appreciated, currently have worked 14 hours straight, but don't want to leave the office while there is a chance of getting this nailed tonight. (Also have an out of hours support call to do on the way home, and my wife and son are already fast asleep and my daughter is on a trip to France so no desperate rush to get home :-/ )

[djprice]

Sorry forgot, above relates to a fresh campaign and fresh lists