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Messages - cfernandes

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31
Users / Re: Asterisk + 10.04 + Dahdi + TDM800p
« on: May 06, 2013, 01:06:45 pm »
Like Posde said ,

no support for this hardware but , you can manualy configure  this .

basicaly you need to

edit /etc/asterisk/chan_dahdi.conf, and assign a specific context to our channel(s)

like this

[channels]
language=pt_BR
context=from-trunk
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
channel=2 


32
Users / Re: Asterisk + 10.04 + Dahdi + TDM800p
« on: May 05, 2013, 01:25:41 am »
a think that you need to install  asterisk-dahdi 

33
Installation issues / Re: Cisco 7970 - upgrading chain
« on: March 24, 2013, 10:35:44 pm »
i use firmware version 9.3.1  if you need you can download from 

  http://www.io.inf.br/linuxmce/cmterm-7970_7971-sccp.9-3-1-1.zip

34
Users / Re: Asterisk CDR registering error.
« on: February 27, 2013, 10:08:29 pm »
try  with  my   

35
Users / Re: Asterisk CDR registering error.
« on: February 26, 2013, 11:43:15 am »
my cdr is correct   check   on  db_phone_config.sh   

if look like that
  LINESSQL="$LINESSQL INSERT INTO $DB_Extensions_Table (context,exten,priority,app,appdata) VALUES
        ('$context','$phonenumber','1','Set','__FROM_DID=\${EXTEN}'),
        ('$context','$phonenumber','2','Set','PAI=\${SIP_HEADER(P-Asserted-Identity)}'),
        ('$context','$phonenumber','3','Set','PAF=\${SIP_HEADER(FROM)}'),
        ('$context','$phonenumber','4','Set','CURRENT_PAI_LENGTH=\${LEN(\${PAI})}'),
        ('$context','$phonenumber','5','Set','CURRENT_PAF_LENGTH=\${LEN(\${PAF})}'),
        ('$context','$phonenumber','6','gotoif','\$[\${CURRENT_PAI_LENGTH} > 0] ? 11'),
        ('$context','$phonenumber','7','gotoif','\$[\${CURRENT_PAF_LENGTH} > 0] ? 19'),
        ('$context','$phonenumber','8','Noop','Incoming call from \${CALLERID(num)}'),
        ('$context','$phonenumber','9','Set','FAX_RX='),
        ('$context','$phonenumber','10','Goto','custom-linuxmce,$line,1'),
        ('$context','$phonenumber','11','noop','config p asserted id ${PAI}'),
        ('$context','$phonenumber','12','set','tmpcid=\${CUT(PAI,:,2)}'),
        ('$context','$phonenumber','13','Set','tmpcid=\${CUT(tmpcid,@,1)}'),
        ('$context','$phonenumber','14','Set','CALLERID(num)=\${tmpcid}'),
        ('$context','$phonenumber','15','Noop','Incoming call from \${CALLERID(num)}'),
        ('$context','$phonenumber','16','Set','FAX_RX='),
        ('$context','$phonenumber','17','Set','CALLERID(ani)='),
        ('$context','$phonenumber','18','Goto','custom-linuxmce,$line,1'),
        ('$context','$phonenumber','19','noop','config SIP Header From ${PAF}'),
        ('$context','$phonenumber','20','set','tmpcid=\${CUT(PAF,:,2)}'),
        ('$context','$phonenumber','21','Set','tmpcid=\${CUT(tmpcid,@,1)}'),
        ('$context','$phonenumber','22','Set','CALLERID(num)=\${tmpcid}'),
        ('$context','$phonenumber','23','Noop','Incoming call from \${CALLERID(num)}'),
        ('$context','$phonenumber','24','Set','FAX_RX='),
        ('$context','$phonenumber','25','Set','CALLERID(ani)='),
        ('$context','$phonenumber','26','Goto','custom-linuxmce,$line,1');"

Code: [Select]
| calldate            | clid           | src            | dst | dcontext          | channel              | dstchannel | lastapp    | lastdata | duration | billsec | disposition | amaflags | accountcode | userfield | uniqueid       |
+---------------------+----------------+----------------+-----+-------------------+----------------------+------------+------------+----------+----------+---------+-------------+----------+-------------+-----------+----------------+
| 2013-02-25 12:28:52 | 18021382320001 | 18021382320001 | s   | from-sip-external | SIP/0.0.0.0-00000060 |            | Congestion | 5        |       13 |      13 | ANSWERED    |        3 |             |           | 1361806132.367 |
| 2013-02-25 12:18:53 | 13021382320001 | 13021382320001 | s   | from-sip-external | SIP/0.0.0.0-0000005f |            | Congestion | 5        |       13 |      13 | ANSWERED    |        3 |             |           | 1361805533.366 |
| 2013-02-25 12:08:54 | 15021382320001 | 15021382320001 | s   | from-sip-external | SIP/0.0.0.0-0000005e |            | Congestion | 5        |       13 |      13 | ANSWERED    |        3 |             |           | 1361804934.365 |
+---------------------+----------------+----------------+-----+-------------------+----------------------+------------+------------+----------+----------+---------+-------------+----------+-------------+-----------+----------------+

36
Users / Re: 10.04 Telecom problem
« on: February 24, 2013, 03:41:24 pm »
you can post  asterisk logs ?

and result off
this query

use asterisk
select * from extensions where context like 'outbound-allroutes%';

37
Users / Re: 10.04 Telecom problem
« on: February 24, 2013, 12:26:05 am »
I'm working on it

38
Users / Re: 10.04 Telecom problem
« on: February 21, 2013, 06:49:58 pm »
i have 3 lines  configured  and  i use  "0"   for the first line  , 99  for the second  and  88 for the third

39
Users / Re: 10.04 Telecom problem
« on: February 21, 2013, 06:24:22 pm »
 the current  dialing rules, need a prefix to indicate which route should be used.


40
Users / Re: 10.04 Telecom problem
« on: February 21, 2013, 06:10:02 pm »
you need to configure  a prefix  to dial out.


41
Users / Re: 10.04 Telecom problem
« on: February 21, 2013, 10:55:54 am »
willow3

i try to help you.

how your  line is configured .

carlos

42
Users / Re: detection key of phone does not work after calling behind
« on: January 14, 2013, 11:58:34 am »
you need  send more info

open asterik cli 
with
asterisk -r -vvvvvvvvvv

and try a call  then post logs

43
Users / Re: IVR after fresh install
« on: January 12, 2013, 04:55:39 pm »
configure on lmce-admin  telecom default language english and try to generate ivr  again

44
Users / Re: Change default phone menu
« on: January 12, 2013, 12:48:10 pm »
you checked the menu telecom call routing, to verify that it meets the options?

45
Users / Re: Asterisk Security
« on: January 12, 2013, 12:44:26 pm »
you no need to change sip.conf

this change is implemented on asterisk realtime database  by Foxi.


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