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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: peterpan746 on June 09, 2008, 04:26:44 PM

Title: Extensions can't dail each other
Post by: peterpan746 on June 09, 2008, 04:26:44 PM

First off I'm new to both SME and Asterisk, but you have to start somewhere.

I have installed Asterisk 1.4 on SME Server 7.3 (Setup as server only)
I'm using the "free" version of X-Lite soft phones installed on two pc's on the same network.
The soft phones register fine, and when I attempt to make a call it shows in the FOP(Operator Panel) that the phone/extension is being used.

Now when I try to call from extension 5000 to 5001 I get the voice mail greeting.  I tried to search the forum, but seems like my wording in the search phrase isn't matching anything.

Any help will be appreciated.

P.S.  If I click on the extensions tab in SAIL is shows the state of the extension as an "X"

Title: Re: Extensions can't dail each other
Post by: SARK devs on June 10, 2008, 06:39:13 AM

Hello,

Difficult to diagnose without more info.  However...  Start by watching the asterisk log to see what is happening during the call attempt.  If it isn't obvious why the call is failing then post the log here.

Also if SAIL shows the extension status as an X then it probably isn't truly registered.  Do sip show peers at the asterisk console and post the output here.

Kind Regards

S

 

Title: Re: Extensions can't dail each other
Post by: peterpan746 on June 10, 2008, 09:04:08 AM

With the help of a friend and the Darrel May Port scanner it seems that port 5060 isn't open on the SME server:

Starting nmap 3.70 ( http://www.insecure.org/nmap/ ) at 2008-06-10 08:56 SAST
Interesting ports on localhost (127.0.0.1):
(The 1643 ports scanned but not shown below are in state: closed)
PORT     STATE SERVICE
22/tcp   open  ssh
25/tcp   open  smtp
26/tcp   open  unknown
80/tcp   open  http
110/tcp  open  pop3
139/tcp  open  netbios-ssn
143/tcp  open  imap
389/tcp  open  ldap
443/tcp  open  https
465/tcp  open  smtps
515/tcp  open  printer
548/tcp  open  afpovertcp
980/tcp  open  unknown
993/tcp  open  imaps
995/tcp  open  pop3s
2000/tcp open  callbook
3128/tcp open  squid-http

When trying to setup a call I get this:

 -- Executing [5001@internal:1] AGI("SIP/5000-08970d58", "selintra|OutCluster|5001") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [5001@default:1] AGI("SIP/5000-08970d58", "selintra|InCall|") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (SIP/5001|15|tT)
[Jun 10 08:59:05] WARNING[5134]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- AGI Script Executing Application: (Background) Options: (silence/1)
    -- <SIP/5000-08970d58> Playing 'silence/1' (language 'en')
    -- AGI Script Executing Application: (Voicemail) Options: (5001|u)
    -- <SIP/5000-08970d58> Playing 'vm-theperson' (language 'en')
    -- <SIP/5000-08970d58> Playing 'digits/5' (language 'en')
  == Spawn extension (default, 5001, 1) exited non-zero on 'SIP/5000-08970d58'
    -- Executing [h@default:1] Hangup("SIP/5000-08970d58", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/5000-08970d58'
[Jun 10 08:59:23] NOTICE[4672]: chan_sip.c:7511 sip_reg_timeout:    -- Registration for '65xxxx@fwd.pulver.com' timed out, trying again (Attempt #8055)
    -- ast_get_srv: SRV lookup for '_sip._udp.fwd.pulver.com' mapped to host fwd.pulver.com, port 5060
[Jun 10 08:59:24] NOTICE[4672]: chan_sip.c:12521 handle_response_register: Failed to authenticate on REGISTER to '65xxxx@fwd.pulver.com' (Tries 3)


Sip show peers - Result

kronos*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
909715/peterpan74          69.90.155.70                5060     OK (354 ms)
5002/5002                  (Unspecified)    D          0        UNKNOWN
5001/5001                  (Unspecified)    D          0        UNKNOWN
5000/5000                  (Unspecified)    D          0        UNKNOWN
4 sip peers [Monitored: 1 online, 3 offline Unmonitored: 0 online, 0 offline]

Title: Re: Extensions can't dail each other
Post by: SARK devs on June 10, 2008, 12:59:25 PM

Hello

SAIL sets the SME database to automatically open 5060.  You can check this by doing  config show SIP.  It should look like this...

Code: [Select]

~]# config show SIP
SIP=service
    UDPPort=5060
    access=public
    status=enabled

If you still believe that the port is closed then you can temporarily turn the SME firewall off (in order to test your suspicions) with

Code: [Select]

/etc/init.d/masq stop
To turn it back on do

Code: [Select]

/etc/init.d/masq start
You may also want to install wireshark so you can watch the packets arriving and departing...

Code: [Select]

yum install wireshark --enablerepo=base
you can watch your softphones attempting to register by doing

Code: [Select]

tethereal -R sip -f "host {ip-of-your-softphone}"
Kind Regards

S

Title: Re: Extensions can't dail each other
Post by: peterpan746 on June 11, 2008, 12:31:36 PM

Hi there,

Thank you for all the help, but sadly it was my own incompetence that led to the fault.  I didn't setup the X-Lite phone correctly ( The tick at "Register with domain and receive incoming calls"  and "Send outbound via").

Once again thank you for you help, and hopefully this might help other people that is new.