16
Users / Re: 10.04 Telecom problem
« on: April 21, 2013, 02:23:14 am »I'm working on it
Hi Carlos,
news about the dialplan?
BR,
Paulo
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Hi Carlos,
news about the dialplan?
BR,
Paulo
18
Users / Asterisk SCCP 7970 Buttons.
« on: March 03, 2013, 09:36:19 pm »
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
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
19
Users / Re: Asterisk AST_CONFIG - lost entries
« on: March 03, 2013, 12:10:44 pm »
Done. Opened a ticket with feature_request.
20
Users / Re: Asterisk CDR registering error.
« on: March 01, 2013, 11:56:19 pm »
Thx J and Carlos,
worked like a charm. Now, registering the incoming calls correctly.
Thx again.
worked like a charm. Now, registering the incoming calls correctly.
Thx again.
21
Users / Asterisk AST_CONFIG - lost entries
« on: March 01, 2013, 11:54:58 pm »
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.
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.
22
Developers / Re: Missing asterisk directives.
« on: March 01, 2013, 11:18:40 pm »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]83 0 18 0 sip.conf general alwaysauthreject yesThe 83 is very inportant for fail2ban and 85-88 to have the NAT.
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
I also would like to know which are the rules for cat_metric and var_metric.
Anyway, my sip line register, but i keep gettingCode: [Select]> 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.
> 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'
Way to solve it?
TIA
After updating asterisk, all the entries i did were gone
No more NAT, externip, etc....
23
Users / Re: Asterisk CDR registering error.
« on: March 01, 2013, 12:35:39 am »
apt-get install --reinstall lmce-asterisk done. What's next?
24
Users / Re: Asterisk CDR registering error.
« on: February 28, 2013, 10:47:39 am »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.
25
Users / Re: Asterisk CDR registering error.
« on: February 27, 2013, 08:45:01 pm »
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.
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.
26
Users / Asterisk CDR registering error.
« on: February 26, 2013, 12:23:34 am »
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.
TIA
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]
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
27
Users / Re: Asterisk Security
« on: February 23, 2013, 11:15:25 pm »
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.
For the ones using lmce 1004, please update the fail2ban configuration file according the the wiki example.
For lmce 810, no change is needed.
28
Users / Re: 10.04 Telecom problem
« on: February 23, 2013, 11:02:33 pm »
I also have 3 lines configured, but would be nicer to have a way to define a dialplan, instead of having 3 dialout prefixes .
.
29
Users / Re: Asterisk Security
« on: January 13, 2013, 11:40:07 pm »so I failed on the second step:You can enter it in the asterisk database, table ast_config, mine looks like:
/etc/asterisk/sip.conf doesn't exist
Edit Delete 83 0 18 0 sip.conf general alwaysauthreject yes
30
Users / Re: [SOLVED] Voicemail in 1004
« on: January 11, 2013, 08:46:09 pm »
I remember i corrected the permissions twice after install and updates.