vicidial.org

Hello.

I've been seting up a ViciDialNow installation on a box with a rather specific network configuration: The LAN on which the softphones are and the dedicated WAN my VoIP provider provides are distinct. So my box has two IPs, one on each L/WAN.

The Asterisk setup works, everything works, except that the information of the dialed persons do not show up on the agent's webpage.

I'm wondering to which extent this double IP setup could affect the Perl scripts, and more specifically the perl script feeding back this information from Asterisk to the call-center agent's web page.

Guillaume.

The IP addresses of your VicidialNOW server has nothing to do with the information of the dialed person. Are you doing manual dial or auto-dial?

Manual dial seems to work fine. It's with auto-dial that I'm having this issue of client information not showing up.

I just wasn't that this double IP setup had something to do with this, but auto-dial was working fine when I was using a single remote VoIP provider on the Internet. So I thought this was an issue.

Could you please detail me the flow of information/computation from the point when Asterisk notifies that a call has been answered to the point where the agent's screen is updated?

Thanks for your answer.

Guillaume.

The SIP provider I'm on is not a registrar. No registration required. And auto-dial doesn't work. It used to work when we had a registrar as VoIP provider. Maybe that could be related to my issue?

I've tried to connect to my old SIP provider. That means a single IP setup and a REGISTRY string.

Predictive auto-dialing worked fine.

Tried back my new setup, with 2 IPs and a non-registrar SIP provider.

Manual dialing works, but not auto-dialing.

Where should I look into in order to debug this?

Seen in /var/www/agc/conf_exten_check.php

# This script is designed purely to send whether the meetme conference has live channels connected and which they are
# This script depends on the server_ip being sent and also needs to have a valid user/pass from the vicidial_users table

So there may be an issue with this $server_ip variable, perhaps... Please help me out with this one...

Have you run "/usr/share/astguiclient/ADMIN_update_server_ip.pl" when you changed the IP addresses of your server? This will update the IP address entries in your Vicidial tables. I think this is what you're referring to.

Yes, I did, an quite a while ago. I set it to the IP visible from the LAN where the SIP softphones are. I did not set it to the IP visible from my VoIP provider.

I'd really like to debug the beast. How can I get started doing so? Who is responsible for feading data to conf_exten_check.php? Fetched from the DB? If so, which tables should I monitor?

Thanks a lot for your answers.

Guillaume.

Sorry, I missed this one. Could you post your Asterisk CLI when dialing? Most SIP carriers I use don't require registrations. Have you tried using the "sip-silence" playback dialplan entry?

yziquel wrote:The SIP provider I'm on is not a registrar. No registration required. And auto-dial doesn't work. It used to work when we had a registrar as VoIP provider. Maybe that could be related to my issue?

sip show peers gives:

101/101 192.168.XXX.XXX D 38742 Unmonitored
100/100 192.168.XXX.XXX D 31732 Unmonitored
cablecom 62.2.XXX.XXX. A 5060 Unmonitored
53 sip peers [53 online , 0 offline]

I should mention that, having had a look at the database, the predictive calls are indeed made and answered, but they are not forwarded properly to the agent: the agent's phone rings, but no voice goes through. And the agent's web page is not updated.

Again manual dial works fine.

I do not what the sip-silence playback entry is about, but looking at the CLI output, it seems that I'm using it.

That's what's on the CLI:

> Channel SIP/100-082263a8 was answered.
-- Executing MeetMe("SIP/100-082263a8", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing NoOp("Local/XXXXXXX514@default-f0b4,2", "") in new stack
-- Executing Dial("Local/XXXXXXX514@default-f0b4,2", "SIP/cablecom/XXXXXXX514") in new stack
-- Called cablecom/XXXXXXX514
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing NoOp("Local/XXXXXXX521@default-13e1,2", "") in new stack
-- Executing Dial("Local/XXXXXXX521@default-13e1,2", "SIP/cablecom/XXXXXXX521") in new stack
-- Called cablecom/XXXXXXX521
-- SIP/cablecom-08229aa8 answered Local/XXXXXXX521@default-13e1,2
> Channel Local/XXXXXXX521@default-13e1,1 was answered.
-- Executing Playback("Local/XXXXXXX521@default-13e1,1", "sip-silence") in new stack
-- Playing 'sip-silence' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, XXXXXXX521, 2) exited non-zero on 'Local/XXXXXXX521@default-13e1,2'
-- Executing DeadAGI("Local/XXXXXXX521@default-13e1,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----8-----0") in new stack
-- Executing AGI("SIP/cablecom-08229aa8", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AGI("SIP/cablecom-08229aa8", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---8-----0 completed, returning 0
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- SIP/cablecom-08214c90 is ringing
-- SIP/cablecom-08214c90 answered Local/XXXXXXX514@default-f0b4,2
> Channel Local/XXXXXXX514@default-f0b4,1 was answered.
-- Executing Playback("Local/XXXXXXX514@default-f0b4,1", "sip-silence") in new stack
-- Playing 'sip-silence' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, XXXXXXX514, 2) exited non-zero on 'Local/XXXXXXX514@default-f0b4,2'
-- Executing DeadAGI("Local/XXXXXXX514@default-f0b4,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----13-----0") in new stack
-- Executing AGI("SIP/cablecom-08214c90", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --13-----0 completed, returning 0
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AGI("SIP/cablecom-08214c90", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
== Spawn extension (default, 8368, 3) exited non-zero on 'SIP/cablecom-08229aa8'
-- Executing DeadAGI("SIP/cablecom-08229aa8", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, 8368, 3) exited non-zero on 'SIP/cablecom-08214c90'
-- Executing DeadAGI("SIP/cablecom-08214c90", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

Asterisk version?
Vicidial/Astguiclient version?

Asterisk CLI looks normal. Have you checked your firewall settings?

Asterisk 1.2.30.2
ViciDial is 1.2 I guess.

There is NO firewall at all!

1 IP is on a LAN (the one is update_server_ip), where the softphones are.
The other IP (the one in 'externip' in sip.conf) is on another private LAN belonging to my VoIP provider. (The Asterisk has two IPs and it belongs to 2 DISTINCT LANs)

The only thing a firewall has to do with all the networking setup is being a gateway for an NTP server, and only that. No Internet, nothing at all. No firewall!!!

I recall that it works when I do it manually. I also recall that if I change my VoIP provider to another one (with a firewall, and only one IP), auto dialing works fine.

I have just tried something else: putting ViciDial on only one IP, and having a second Asterisk server with a double IP (I cannot really get over this double IP stuff) to relay all calls between ViciDial and my VoIP provider.

Same thing. Manual dialing works fine. Automatic Dialing doesn't work.

So my question: given that ViciDial can successfully send the call, where is the piece of code in charge of linking the call to the agent's softphone?

I've tried it today in manual dialing. Agents have four calls simultaneously on their phones. It also seems that ViciDial is completely confused as to which meetme conference rooms it should allocate...

ok, sorry i'm late, and i've read all the above, but i'm a little confused.

is it now that your auto-dial simply doesn't work when you are in "two ip" mode or that the auto-dial works, but the information for the dialed prospect does not appear?

primarily, you want to ignore the web ip as far a vicidial settings are concerned, as the apache server will resolve that without any problems, the vicidial ip is for routing calls. we've got this setup working in several places on several occasions for several reasons.

also please post your VicidialNOW version, your vicidial version and build (NOT the same as your vicidialnow version, they are separate) your asterisk version and your zaptel version and any other software you have installed and any special hardware (such as digium/sangoma cards) and your CPU.