22 thoughts on - (no subject)

  • Hi sam,

    Have solved the problem with your advice. Call drop in 10 seconds without disconnecting a-party call. Thank you very much.

    [TB]

    exten =>_X.,1,Wait(${INCOMING_WAIT})

    exten =>_X.,2,Verbose(TB)

    exten =>_X.,3,Answer()

    exten =>_X.,4,Set(mainLoop=0)

    ;exten =>_X.,5,Set(TIMEOUT(absolute)=50000)

    exten =>_X.,5,Playback(/var/callagent/prompts/monitor/thanks)

    exten => _X.,6,Dial(DAHDI/7/

    09501032209,100,L(30000[:10000][:3000])g)

    exten =>_X.,7,Noop(${DIALEDTIME})

    exten =>_X.,8,Goto(TB,_X.,1)

    exten =>_X.,n,Hangup()

    Cheers
    Vinod Dharashive
    Sent from BlackBerry® on Airtel

  • Hi, I’m not able to configure 8 port card whenever I configure it is showing fatal: error inserting wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown symbol in module, or unknown parameter

    Please help.

    Regards Akhilesh

  • It sounds as though you need to recompile DAHDI-Linux. (Did you compile it before you acquired this card?) Just download the latest DAHDI package Source Code, and compile and install it.

    If you didn’t compile your own kernel from Source Code, then you will also need the package “kernel-devel” (on Fedora / CentOS) or “linux-headers” (on Ubuntu).

  • Basic Dial Plan

    Why is this plan not engaging the line exten => 105,n,Dial(SIP/voipvoip.com/17035013333)
    and dialing the 703 number?

    The logs and debug dont show any problems….

    [incoming]
    exten => 4444444444,1,Answer()
    exten => 4444444444,n,Wait(1)
    exten => 4444444444,n,Playback(beep)
    exten => 4444444444,n,Goto(105,105,1)
    ;
    ;
    [105]
    exten => 105,1,Wait(2)
    exten => 105,n,Playback(hello-world)
    exten => 105,n,Dial(SIP/voipvoip.com/17035013333)
    exten => 105,n,Hangup()

  • Have you included the [105] context within the default context for the extension from which you are dialling 105?

    If 4444444444 from the outside world is failing to trigger it, then it’s possible that Asterisk is seeing the first 105 in “Goto(105,105,1)” as a priority rather than a context,extension,priority. Rename the [105] context to start with a letter.

  • hello all, i want to have ooh323 connection between asterisk and cisco. in my scenario, asterisk is gateway and cisco is gatekeeper. this is my ooh323.conf file:
    [general]
    port20
    bindaddr2.168.0.227
    gateway=yes faststart=yes h245tunneling=yes h323id=gw10@test.com settracelevel
    gatekeeper2.168.0.212
    context=from-trunk disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833

    with this config, gateway is registered in cisco gatekeeper correctly. but when i want to call from it, cisco reject my gateway and h225 asn1 messages say “incomplete address”. i searched a lot and understand that, if a cisco router acts as gateway, it sends h323-id as well as dialed number for gatekeeper but my gateway(which is asterisk), only send dialed number. therefore cisco gatekeeper doesn’t know how route this call and reject it. if i define e164 number in ooh323.conf file, every thing is ok and call routed correctly.

    my question is: can asterisk work with cisco gatekeeper just by h323-id? if yes, how i can do this? in the other words, is it necessary to define e164
    number in ooh323.conf file to have a correct connection or not?

    thanks in advance SAM

  • hello list,

    i have asterisk 1.4 installed i use MixMonitor to record all the inboud calls with the code below my question how can i do to save alse the sip extenssion 223

    exten => 529,1,Answer()
    exten => 529,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
    exten => 529,n,Dial(SIP/223)
    exten => 529,n,Hangup()

    thanks and regards

  • Define it as a variable, use the variable to define the filename….

    Ex.

    exten =>
    529,n,Set(monfile=num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID})

    exten => 529,n,MixMonitor(/var/spool/disa/${monfile}.wav,,)
    hello list,

    i have asterisk 1.4 installed i use MixMonitor to record all the inboud calls with the code below my question how can i do to save alse the sip extenssion 223

    exten => 529,1,Answer()
    exten => 529,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
    exten => 529,n,Dial(SIP/223)
    exten => 529,n,Hangup()

    thanks and regards

  • thanks for your response

    with the code below i can’t get the extenssions 223

    exten => 529,1,Answer()
    exten =>
    529,n,MixMonitor(test_num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID}.wav|av(0)V(0))
    exten => 529,n,Dial(SIP/223)
    exten => 529,n,Hangup()

    i can get my number only with uniqueid

    test_num-0661xxxxxx_name-_529_UID-1376564701.1204.wav

    any help please

    thanks and regards

    2013/8/13 Positively Optimistic

  • Hi

    I am running following asterisk installed with apt on Debian 7.1.

    dpkg -l |grep asterisk ii asterisk 1:1.8.13.1~dfsg-3+deb7u1
    amd64 Open Source Private Branch Exchange (PBX)
    ii asterisk-config 1:1.8.13.1~dfsg-3+deb7u1
    all Configuration files for Asterisk ii asterisk-core-sounds-en-gsm 1.4.22-1
    all asterisk PBX sound files – en-us/gsm ii asterisk-modules 1:1.8.13.1~dfsg-3+deb7u1
    amd64 loadable modules for the Asterisk PBX

    If the incoming INVITE has the following two multiple bodies then it would not respond to that. It won’t even send a Trying. We are using* TCP *only.

    Content-Type: application/sdp
    …. Content-Type: application/ISUP;base=itu-t92+;version=itu-t92+.

    Is this is a known issue? Are later version of asterisk able to deal with such multi-bodies INVITE? I got to play early media so it needs to make some sense out of first SDP.

    Best regards, Adnan

  • To Jonas:

    I have an asterisk box at home and I have this line in my rtp.conf file:

    rtpstart000
    rtpend100

    And My FW is setup to forward all incoming ports of range 10000-10100 to the asterisk PC. I’ve never had a problem since one year, but I have never received more than two simultaneous calls with SIP clients.

    Message: 5
    Date: Fri, 13 Sep 2013 11:49:59 +0200
    From: Jonas Kellens
    Subject: Re: [asterisk-users] RTP port ranges To: Andrew Colin
    Cc: Asterisk Users Mailing List – Non-Commercial Discussion

    Message-ID: <5232DFC7.2030601@telenet.be>
    Content-Type: text/plain; charset=”iso-8859-1″; Format=”flowed”

    Hello,

    and when I define 11500 – 11954 it should use a random port in this range.

    Where is it stated that you MUST use 10000-20000 ???

    Someone else please ?

    Jonas.

  • Hi, all

    I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it “Asterisk11”. I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL’s cdr table(in database “mydatabase”) via cdr_adaptive_odbc. The “SIP/A221” is another asterisk machine named it “Elastix24”.

    I have two BIG QUESTIONs about cdr_adaptive_odbc.

    First, I have answered call from Elastix24 and I can listen the music file played from Asterisk11. In another word, this call should be answered and its billsec is greater than 0.

    Second, if I don’t want to use forkcdr(), how to config it and I can get another cdr record that call from SIP/A221(Elastix24) to my Exten:777777?

    I know that the outgoing file will make a call to Local Channel and try to Dial SIP/A221. If it answered, this old channel should be hangup and generate another new channel to connect to Extension:777777(my callback exten).

    I can’t find two cdr records in mycdr table. mysql> select * from gvl_cdr;
    +———————+——+—–+——-+———————+—————————————————+——————-+———+————————–+———-+———+————-+———-+————-+———–+————–+————–+———-+————-+———+———-+——–+
    | calldate | clid | src | dst | dcontext | channel
    | dstchannel | lastapp |
    lastdata | duration | billsec | disposition | amaflags |
    accountcode | userfield | uniqueid | linkedid | sequence |
    peeraccount | phoneno | callerid | userid |
    +———————+——+—–+——-+———————+—————————————————+——————-+———+————————–+———-+———+————-+———-+————-+———–+————–+————–+———-+————-+———+———-+——–+
    | 2014-01-08 14:37:01 | | |777777 | from-internal-out-7 |
    Local/777777@from-internal-out-7-00000000;2 | SIP/A221-00000000 |
    Dial | SIP/A221/777777,30 | 17 | 0 | ANSWERED |
    3 | | | 1389163021.1 | 1389163021.0 | 1 |
    | 777777 | | 7 |

    Even I try to add ForkCDR or ResetCDR. The billsec is 0 in other record(the
    3th one). mysql> select * from gvl_cdr;
    +———————+——————–+———+—————+———————+—————————————————-+——————-+———+——————————-+———-+———+————-+———-+————-+———–+————–+————–+———-+————-+———+———-+——–+
    | calldate | clid | src | dst |
    dcontext | channel |
    dstchannel | lastapp | lastdata | duration |
    billsec | disposition | amaflags | accountcode | userfield | uniqueid |
    linkedid | sequence | peeraccount | phoneno | callerid | userid |
    +———————+——————–+———+—————+———————+—————————————————-+——————-+———+——————————-+———-+———+————-+———-+————-+———–+————–+————–+———-+————-+———+———-+——–+
    | 2014-01-08 14:34:04 | | | 777777 |
    from-internal-out-7 | Local/777777@from-internal-out-7-00000000;2 |
    SIP/A221-00000000 | Dial | SIP/A221/777777,30 | 15 |
    0 | ANSWERED | 3 | | | 1389162844.1 |
    1389162844.0 | 1 | | 777777 | | 7 |
    | 2014-01-08 14:34:04 | “device” <1000> | 1000 | 777777 |
    from-6 | Local/777777@from-internal-out-7-00000000;1 |
    | ForkCDR | | 20 |
    5 | ANSWERED | 3 | | | 1389162844.0 |
    1389162844.0 | 0 | | 777777 | | 7 |
    | 2014-01-08 14:34:24 | “device” <777777> | 777777 | 777777 |
    from-6 | Local/777777@from-internal-out-7-00000000;1 |
    | Read | CALLBACK,custom-gvl/2,1,s,1,3 | 0 |
    0 | NO ANSWER | 3 | | | 1389162844.0 |
    1389162844.0 | 3 | | | | 0 |

    —————– /var/spool/asterisk/outgoing/777777.call Channel:Local/777777@from-internal-out-7
    WaitTime:30
    Context:from-6
    Extension:777777
    Priority:1
    Set:CLID=
    Set:EXT=777777
    Set:USERID=7

    ————– /etc/asterisk/extensions.conf lists below:
    [from-internal-out-7]
    exten => _X.,1,Set(CALLERID(number)=${CLID})
    exten => _X.,n,Set(CDR(phoneno)=${EXT})
    exten => _X.,n,Set(CDR(userid)=${USERID})
    exten => _X.,n,Set(CDR(callerid)=${CLID})
    exten => _X.,n,Dial(SIP/A221/${EXTEN},30)
    exten => failed,1,Hangup

    [from-6]
    exten => _X.,1,Answer()
    exten => _X.,n,Set(CALLERID(number)=${EXTEN})
    exten => _X.,n,Set(CDR(phoneno)=${EXTEN})
    exten => _X.,n,Set(CDR(userid)=${USERID})
    exten => _X.,n,Set(CDR(callerid)=${CLID})
    exten => _X.,n,Set(RETRYCOUNT=$[ 1])
    exten => _X.,n(countdown),Set(RETRYCOUNT=$[ ${RETRYCOUNT} – 1])
    exten => _X.,n(replay),Read(CALLBACK,custom-gvl/2,1,s,1,3)
    exten => _X.,n,GotoIf($[‘${CALLBACK}’=’0’]?replay:pressother)
    exten => _X.,n,GotoIf($[${RETRYCOUNT} > 0]?countdown:pressother)
    exten => _X.,n(pressother),NoOp(CALLBACK:${CALLBACK})
    exten => _X.,n,Hangup()

    exten => h,1,Hangup()
    exten => i,1,Hangup()

    ————– /etc/asterisk/cdr.conf lists below:
    [general]
    enable=yes unanswered = yes congestion = yes

    [csv]
    usegmtime=no ; log date/time in GMT. Default is “no”
    loguniqueid=yes ; log uniqueid. Default is “no”
    loguserfield=yes ; log user field. Default is “no”
    accountlogs=yes ; create separate log file for each account code. Default is “yes”

    ————– /etc/odbc.ini
    [asterisk-connector]
    Description = MySQL connection to ‘asterisk’ database Driver = MySQL
    Database = mydatabase Server = localhost UserName = root Password = mypassword Port = 3306
    Socket = /var/run/mysqld/mysqld.sock

    ————– /etc/asterisk/res_odbc.conf lists below:
    [ENV]

    [asterisk]
    enabled => yes dsn => asterisk-connector password => mypassword pre-connect => yes sanitysql => select 1
    idlecheck => 30
    connect_timeout => 20

    ————– /etc/asterisk/cdr_adaptive_odbc.conf lists below:
    [cdr]
    connection=asterisk table=cdr alias start => calldate alias phoneno => phoneno alias userid => userid alias callerid => callerid

    ————– asterisk’s CLI:
    ubuntu*CLI> module reload cdr_adaptive_odbc.so
    — Reloading module ‘cdr_adaptive_odbc.so’ (Adaptive ODBC CDR backend)
    == Parsing ‘/etc/asterisk/cdr_adaptive_odbc.conf’: Found
    — Found adaptive CDR table cdr@asterisk.
    — Found alias start for column calldate in cdr@asterisk
    — Found alias phoneno for column phoneno in cdr@asterisk
    — Found alias callerid for column callerid in cdr@asterisk
    — Found alias userid for column userid in cdr@asterisk

    — Attempting call on Local/777777@from-internal-out-7 for 777777@from-6:1
    (Retry 1)
    — Executing [777777@from-internal-out-7:1]
    Set(“Local/777777@from-internal-out-7-00000000;2”, “_FLOWID=6”) in new stack
    — Executing [777777@from-internal-out-7:2]
    Set(“Local/777777@from-internal-out-7-00000000;2”, “CALLERID(number)=”) in new stack
    — Executing [777777@from-internal-out-7:3]
    Set(“Local/777777@from-internal-out-7-00000000;2”, “CDR(phoneno)=777777”)
    in new stack
    — Executing [777777@from-internal-out-7:4]
    Set(“Local/777777@from-internal-out-7-00000000;2”, “CDR(userid)=7”) in new stack
    — Executing [777777@from-internal-out-7:5]
    Dial(“Local/777777@from-internal-out-7-00000000;2”, “SIP/A221/777777,30”)
    in new stack
    == Using SIP RTP CoS mark 5
    — Called SIP/A221/777777
    — SIP/A221-00000000 is ringing
    > 0xb6c04f70 — Probation passed – setting RTP source address to
    192.168.1.226:15396
    [Jan 8 14:37:01] WARNING[9132][C-00000000]: dsp.c:1490 ast_dsp_process:
    Inband DTMF is not supported on codec g729. Use RFC2833
    — SIP/A221-00000000 is ringing
    — SIP/A221-00000000 answered Local/777777@from-internal-out-7-00000000
    ;2
    > Channel Local/777777@from-internal-out-7-00000000;1 was answered
    — Executing [777777@from-6:1]
    Answer(“Local/777777@from-internal-out-7-00000000;1”, “”) in new stack
    — Executing [777777@from-6:2]
    Set(“Local/777777@from-internal-out-7-00000000;1”,
    “CALLERID(number)=777777”) in new stack
    — Executing [777777@from-6:3]
    Set(“Local/777777@from-internal-out-7-00000000;1”, “CDR(phoneno)=777777”)
    in new stack
    — Executing [777777@from-6:4]
    Set(“Local/777777@from-internal-out-7-00000000;1”, “CDR(userid)=7”) in new stack
    — Executing [777777@from-6:5]
    Set(“Local/777777@from-internal-out-7-00000000;1”, “RETRYCOUNT=1”) in new stack
    — Executing [777777@from-6:6]
    Set(“Local/777777@from-internal-out-7-00000000;1”, “RETRYCOUNT=0”) in new stack
    — Executing [777777@from-6:7]
    Read(“Local/777777@from-internal-out-7-00000000;1”,
    “CALLBACK,custom-gvl/2,1,s,1,3”) in new stack
    — Accepting a maximum of 1 digits.
    Playing
    ‘custom-gvl/2.slin’ (language ‘en’)
    > 0xb6c04f70 — Probation passed – setting RTP source address to
    192.168.1.226:15396
    [Jan 8 14:37:23] NOTICE[9132][C-00000000]: res_odbc.c:1524
    odbc_obj_connect: Re-connecting asterisk
    [Jan 8 14:37:23] NOTICE[9132][C-00000000]: res_odbc.c:1559
    odbc_obj_connect: res_odbc: Connected to asterisk [asterisk-connector]
    == Spawn extension (from-internal-out-7, 777777, 13) exited non-zero on
    ‘Local/777777@from-internal-out-7-00000000;2’
    — User disconnected
    — Executing [h@from-6:1] Hangup(“SIP/A221-00000000”, “”) in new stack
    == Spawn extension (from-6, h, 1) exited non-zero on ‘SIP/A221-00000000’
    [Jan 8 14:37:40] NOTICE[9131]: pbx_spool.c:402 attempt_thread: Call completed to Local/777777@from-internal-out-7

  • Dahdi on Archlinux

    I was able to compile the latest 2.9 Dahdi in archlinux on the Beaglebone black without errors. I ran make install and make config.

  • Submission.

    Thanks,

    Francisco Leonardo Mota Analista de Operações DAGSer – Diretoria Adjunta de Gestão de Serviços RNP – Rede Nacional de Ensino e Pesquisa
    Site:http://www.rnp.br Tel.:+55 61 3243-4384
    Cel.:+55 61 9189-6660

  • Hi,

    Im trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is working fine but i cannot dial out. i don’t hear anything on the phone and asterisk CLI also does not show anything. my config is. please advice.

    [2001]
    type=endpoint
    context=out-local
    disallow=all
    allow=ulaw
    allow=alaw
    transport=system-udp
    auth 01
    aors 01
    direct_media=no
    rtp_symmetric=yes
    force_rport=yes
    allow=alaw
    allow=speex
    allow=speex16
    allow=speex32
    allow=gsm

    [2001]
    type=aor
    qualify_frequencyP00
    authenticate_qualify=yes
    max_contacts=1
    remove_existing=yes

    [2001]
    type=auth
    auth_type=userpass
    password=test
    username=test

    Best Regards, Madushan