Messages

I'm working on it

Hi Carlos,
news about the dialplan?
BR,
Paulo

No hints? 

Hya,
I do have 4 phone extensions at home, 2 of them on SCCP channel, with 2 Cisco 7970 and 2 SIP on siphones. I could alter the buttonconfig table, so to make one of the 7970 have 2 line buttons (one for it's extension and the second for the other 7970 extension).
I would like to also have buttons 3 and 4 of this 7970 assigned to the other 2 sip extensions and button 5 to my doorbell?
I did enter the 2 sip extensions in the buttonconfig table, but the 7970 only shows the first 2 buttons assigned to the sccp extensions.
Does anyone knows how to have buttons 3 and 4 assigned to the sip extensons and button 5 to my doorbell?
Thx in advance.
Paulo

Done. Opened a ticket with feature_request.

Thx J and Carlos,
worked like a charm. Now, registering the incoming calls correctly.
Thx again.

Hya,
today i discovered that some directives i've inserted into the asteriks ast_table were lost, after an update (don' t know which).
All the entries regarding NAT, EXTERNIP, LOCALNET and so on, to enable remote extensions simply disapeared.
As i had the external extension working, and after for coincidence updated ios and softphone i did for the last four weeks try to fix the problem on the softphone configs, with not luck, until i looked into the asterisk config to discover my directives were not there.
Would be nice if the updates do not delete the custom entries.
Best regards.

Hya,

installing my sip line (voipcheap), i did note it did not register.
So, looking into the database, table ast_config, i noted that some directives were missing.

Code Select Expand


83 0 18 0 sip.conf general alwaysauthreject yes
85 0 18 0 sip.conf general nat                        yes
86 0 60 0 sip.conf general externhost        mydyndns.homeunix.org
87 0 5 0 sip.conf general externrefresh        5
88 0 60 0 sip.conf general localnet               192.168.80.0/255.255.255.0
89 0 9 0 sip.conf general allow                g729
90 0 10 0 sip.conf general allow                g723
91 0 101 0 sip.conf general register                pwollny:XXXXXXXXX@sip.voipcheap.com/2062036594

The 83 is very inportant for fail2ban and 85-88 to have the NAT.
I also would like to know which are the rules for cat_metric and var_metric.

Anyway, my sip line register, but i keep getting

Code Select Expand


> doing dnsmgr_lookup for 'sip.voipcheap.com'
      > doing dnsmgr_lookup for 'sip.voipcheap.com'
      > doing dnsmgr_lookup for 'sip.voipcheap.com'
      > doing dnsmgr_lookup for 'sip.voipcheap.com'
      > doing dnsmgr_lookup for 'sip.voipcheap.com'

in the asterisk verbose, i can place calls, but no sound incoming and outgoing and receiving a call is rejected with "no service message" in my asterisk.

Way to solve it?

TIA

After updating asterisk, all the entries i did were gone
No more NAT, externip, etc....

apt-get install --reinstall lmce-asterisk done. What's next?

Your db_phone_config.sh is a 5 month old version.  Did apt-get report any packages held back?

J.

No, apt-get report did no report held back.

This is what i have in my db_phone_config.sh. The last apt-get update && upgrade is three days ago.

LINESSQL="$LINESSQL INSERT INTO $DB_Extensions_Table (context,exten,priority,app,appdata) VALUES \
        ('$context','$phonenumber','1','Set','__FROM_DID=\${EXTEN}'), \
        ('$context','$phonenumber','2','Set','PAI=\${SIP_HEADER(FROM)}'),\
        ('$context','$phonenumber','3','gotoif','\$[\"\${PAI}\" = \"\"] ? 4:8'), \
        ('$context','$phonenumber','4','Set','CALLERID(num)=\${CALLERID(ani)}'), \
        ('$context','$phonenumber','5','Noop','Incoming call from \${CALLERID(num)}'), \
        ('$context','$phonenumber','6','Set','FAX_RX='), \
        ('$context','$phonenumber','7','Goto','custom-linuxmce,$line,1'),\
        ('$context','$phonenumber','8','noop','Using p-asserted-id SIP header: ${PAI}'),\
        ('$context','$phonenumber','9','set','tmpcid=\${CUT(PAI,:,2)}'), \
        ('$context','$phonenumber','10','Set','tmpcid=\${CUT(tmpcid,@,1)}'), \
        ('$context','$phonenumber','11','Set','CALLERID(num)=\${tmpcid}'),\
        ('$context','$phonenumber','12','Noop','Incoming call from \${CALLERID(num)}'),\
        ('$context','$phonenumber','13','Set','FAX_RX='), \
        ('$context','$phonenumber','14','Goto','custom-linuxmce,$line,1');"
}

It's different from yours.

Hya,
browsing the Asterisk CDR table, i did note the it registers wrong. Incoming calls coming via my SPA3102 is registered as caller id "Unknown Caller" <2122498618>, which is my POTS number.

Code Select Expand


calldate clid src dst dcontext channel dstchannel lastapp lastdata duration billsec disposition amaflags accountcode userfield uniqueid
2013-02-24 20:46:54 "Unknown Caller" <2122498618> 2122498618 2122498618 from-trunk SIP/2122498618-00000009 Local/204@trusted-cac0;1 Set TIMEOUT(response)=20 41 11 ANSWERED 3     1361749614.37

Is this error known?
TIA

Wiki http://wiki.linuxmce.org/index.php/Fail2ban_-_A_tool_against_brute_force was updated for asterisk 1.8. As asterisk 1.8 logs includes the host IP and the port, fail2ban was not blocking attackers IP.
For the ones using lmce 1004, please update the fail2ban configuration file according the the wiki example.
For lmce 810, no change is needed.

I also have 3 lines configured, but would be nicer to have a way to define a dialplan, instead of having 3 dialout prefixes .

so I failed on the second step:

/etc/asterisk/sip.conf doesn't exist

You can enter it in the asterisk database, table ast_config, mine looks like:
   Edit    Delete    83    0    18    0    sip.conf    general    alwaysauthreject    yes

I remember i corrected the permissions twice after install and updates.