Autodial does not connect to agents due to Local channel

[scottgutman]

I am having the problem resolving the local channel pointer. The agi-VDAD_ALL_outbound.agi kills any call that is local and will not process it. Therefore, it does not transfer the call to the agent.

I did several fresh installs, following the scratch install docs on CentOS 5.3. and the ubuntu install on ubuntu 8.04 server.

Manual dials work fine as well as call recording. If I did not autodial, I would not know there was a problem.

setup:
2+g quad core AMD 64bit, 8gig ram, 300 gig sata drive
tried both ztdummy and x100p card
CentOS 5.3 and ubuntu 8.04 server
Kernel 2.6.18-128.2.1.el5 x86_64
tried both asterisk 1.2.30.2 and 1.4.21.2
Astguiclient 2.0.5 and 2.0.5 Trunk
flags = tTo
codec = ulaw

sip.conf
[general]
externip =XXX.XX.XX.XX

Settings in both general and individual accounts.
nat = yes
canreinvite=no

I tried to leave as much of the default conf's untouched so i would not breakanything. I only made the minimum changes to get it to work.

1. I tried installing most components with yum and compiling the remaining dependencies.
2. I tried installing the base only then compiling everything from scratch.
3. I tried the ubuntu install script (Very well done, server was running within 1.5 hours.)
3. I tried adding multiple sipsilence entries and multiple agi-VDAD_ALL_outbound entries. sip-silence is supposed to force Asterisk to resolve the local channel, but it does not in my case.
4. I changed the playback file to a longer gsm recording in exten 8365.
5. I changed the playback file to a wav file. (a standard file in the sounds dir)
6. I created a t.call file to call my cell and playback a file. I put it in /var/spool/asterisk/outgoing. Asterisk got it and called my cell. I answered the call and the CLI showed that the file was being played. I heard only silence, but the length of time of the silence was about the length of the recording.
7. I tried the exact same procedure, but I called my internal softphone. It worked perfectly.

Can this be a nat issue? if yes, then why does the softphone to asterisk to gafachi work?

It seems that asterisk has no problem passing the sip packets between the xlite softphone and sip.gafachi.com. But it can't initiate packets that originate from itself.

<t.call>
Channel: SIP/19545799999@gafachi
Callerid: 9542000000
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: scott
Extension: 10

<extentions.conf>
[scott]
exten => 10,1,Answer()
exten => 10,n,Wait(1)
exten => 10,n,Playback(vtiger-fax)
exten => 10,n,Playback(agent-loggedoff)
exten => 10,n,Wait(1)
exten => 10,n,Hangup()

I don't want to use the quick fix, cuz i will forget about it and some how it will come to bite me in the @#$.

Does anyone have any ideas?

[lerroux]

what the dialplan context you use for dialing? dont forget to use: oTt...

[scottgutman]

I am using the default context.

<from entensions-vicidial>
; VICIDIAL Carrier: gafachi - gafachi
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(SIP/gafachi/${EXTEN:1},,tTo)
exten => _91NXXNXXXXXX,3,Hangup

I am in process of getting a few static ip's and will try putting the asterisk box directly on the net to take NAT out of the picture. How do you recommend securing the box? iptables?

[lerroux]

iptables would be fine. try looking at the logs, see if you can come up with anything that's weird and post it here.

[mflorell]

Have you tried another carrier?

[scottgutman]

Yes, I tried both proximiti and gafachi. They both had the same response.

actually, i did a few tests with the dynamic ip assigned by the isp.

It works, I am waiting to report it when i am 100% sure. I get my static ip's tomorrow, but won't be able to run a full scale test until tuesday.

Thank you for checking up. I appreciate it.

I still don't understand why asterisk is having a problem resolving the channel while nat is happening, if it could route the sip packets between Voip provider and softphone.
Actually, it was double natting, as the modem was doing it's own natting, plus the router. I guess it truely got mangled.

Any way public ip's will fix that. no nat. Chaulk up anther reason why the local channel won't resolve.

I am worried about securing the box against intruders.

I will report back in a few days.

[scottgutman]

Well, we have been dialing for 2 days with public IP's so it seems the local channel resolution problem has been fixed.

[spinto]

I'm having the same problem. Are you saying you resolved it by connecting directly to the internet (bypassing the router/firewall)? What type of firewall do you have?

[scottgutman]

Exactly!!

As an added benefit, all the agents said that the calls sounded "crisper and clearer."

I am using iptables that I configured with firewall builder. I put in a second nic in the dialer to connect to the internet. When I created my firewall rules, the only inbound traffic allowed is SIP and RTP. Everything else is denied on that interface. Then I allowed all internal traffic full access to the internal nic and denied it to the external nic to prevent spoofing.

I bypassed 2 routers/firewalls doing NAT with this procedure. The first was the Comcast SMC Router. It did NAT to the DMZ network and forwarded all ports to the internal router.

Then the second router was a CentOS5.3 box running iptables and dhcpd. This box did NAT from the DMZ to the internal network.

I assume that somewhere in that path, the sip or rtp packets got slightly corrupted. I say slightly, because phone communication worked fine. Asterisk's inability to resolve the local pointer was the only problem.

If someone could point out any flaws in my logic, I would like to know. How does Asterisk decide if a call is local or sip??? Does it lookup the remote ip address and see if it is local?