system toolbox » voip

pozive mogu primati

http://www.voipuser.org/forum_topic_8094.html

Posted: Dec 19, 2006 - 11:59 AM     Reply with quote Back to top
hi all

i just cant make it work.. i´ve a billion pci card (BRI S0) with trixbox.

i´ve install-zaphfc, the card was detected. so far so god.

wen i try to place a call, i get the message " All Circuits are busy " i cant figure it out

here is my logs... thanks for any help

[root@asterisk1 ~]# nano /etc/asterisk/zapata.conf

[trunkgroups]

[channels]

language=pt
context=from-zaptel
signalling=bri_cpe_ptmp
rxwink=350 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include genzaptelconf configs
#include zapata-auto.conf

group=1

;Include AMP configs
#include zapata_additional.conf

;Include BRI-HFC configs
#include zapata-BRI-HFC.conf

-------------------------------------------------------------

[root@asterisk1 ~]# cat /proc/zaptel/1
Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 DEACTIVATED (F5)" AMI/CCS

1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)

i´ve no line connect, and it says (in use) :\

------------------------------------------------------------

asterisk1*CLI> zap show status
Description Alarms IRQ bpviol CRC4
HFC-S PCI A ISDN card 0 [TE] layer 1 DE OK 0 0 0

-------------------------------------------------------------

thanks again for any help

regards

http://www.voip-info.org/wiki/view/Asterisk+zaphfc

...

Problem with ZapHFC with ISDN BRI HFC cards and signalization (busy etc.)

    * edit /etc/asterisk/zapata.conf and insert the line for correct ISDN signalization: 

priindication=outofband

obori asterisk ukloni module

root@ifold:~# /etc/init.d/asterisk stop
Stopping Asterisk PBX: .
root@ifold:~# modprobe -r vzaphfc
root@ifold:~# modprobe -r zaptel
root@ifold:~# modprobe vzaphfc

root@ifold:~# cat /proc/zaptel/1

Span 1: ZTHFC1 "HFC-S PCI A Zaptel Driver card 0 [TE]" (MASTER) 

       1 ZTHFC1/0/1 
       2 ZTHFC1/0/2 
       3 ZTHFC1/0/3

root@ifold:~# /etc/init.d/asterisk start
Starting Asterisk PBX: asterisk.

dodao sam u zapata.conf

priindication=outofband

ali sada asterisk uopšte ne podiže zap interfejsa ?!

idem restartovati ifold

1. uspio pozvati fiksnu liniju: 212643
2. nisam uspio pozvati mobitel
3. asterisk prima pozive moj_mobitel => officesa

ulovio sam šta se desi kada uradim modprobe -r vzaphfc i modprobe zaphfc

  == Parsing '/etc/asterisk/zapata.conf': Found
[Aug 29 09:48:41] WARNING[8296]: chan_zap.c:1082 zt_open: Unable to specify channel 1: No such device or address
[Aug 29 09:48:41] ERROR[8296]: chan_zap.c:7514 mkintf: Unable to open channel 1: No such device or address
here = 0, tmp->channel = 1, channel = 1
[Aug 29 09:48:41] ERROR[8296]: chan_zap.c:12312 build_channels: Unable to register channel '1-2'

hah vidi ovo

zovem 217956 i zvoni

  -- Executing [217956@demo:1] NoOp("SIP/20-081d6840", ""lokal 6-cif"") in new stack
    -- Executing [217956@demo:2] Macro("SIP/20-081d6840", "set_caller_id") in new stack
    -- Executing [s@macro-set_caller_id:1] NoOp("SIP/20-081d6840", "channel = SIP/20-081d6840") in new stack
    -- Executing [s@macro-set_caller_id:2] GotoIf("SIP/20-081d6840", "?3:5") in new stack
    -- Goto (macro-set_caller_id,s,5)
    -- Executing [s@macro-set_caller_id:5] Set("SIP/20-081d6840", "CALLERID(all)=officesa<269291>") in new stack
    -- Executing [s@macro-set_caller_id:6] NoOp("SIP/20-081d6840", "Finish if-set_caller_id-2") in new stack
    -- Executing [217956@demo:3] Dial("SIP/20-081d6840", "zap/g1/033217956|400|tT") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g1/033217956
    -- Zap/1-1 is proceeding passing it to SIP/20-081d6840
    -- Zap/1-1 is ringing

033217956

    -- Goto (macro-set_caller_id,s,5)
    -- Executing [s@macro-set_caller_id:5] Set("SIP/20-081d6840", "CALLERID(all)=officesa<269291>") in new stack
    -- Executing [s@macro-set_caller_id:6] NoOp("SIP/20-081d6840", "Finish if-set_caller_id-2") in new stack
    -- Executing [033217956@demo:2] Goto("SIP/20-081d6840", "sw-1-03321|10") in new stack
    -- Goto (demo,sw-1-03321,10)
    -- Executing [sw-1-03321@demo:10] Dial("SIP/20-081d6840", "zap/g1/sw-1-03321|400|tT") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g1/sw-1-03321   <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< pa on ne stigne da izvrti sve cifre !
    -- Channel 0/1, span 1 got hangup, cause 41
    -- Zap/1-1 is circuit-busy
    -- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [sw-1-03321@demo:11] Goto("SIP/20-081d6840", "_0.|3") in new stack
    -- Goto (demo,_0.,3)
    -- Executing [_0.@demo:3] NoOp("SIP/20-081d6840", "Finish switch-out-1") in new stack

http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf

• busydetect: If enabled, Asterisk will analyze the audio coming in on the line during a call or a dial attempt to attempt to recognize busy signals. This is useful on analog trunk interfaces both to detect a busy signal when dialing out, and for detecting when the person has hung up. See also Disconnect Supervision. Be sure that you don't use this on digital interfaces like QuadBri cards and so on. Otherwise you will run in "broken calls" problems. default=no
  • busydetect=yes
• busycount: This option requires busydetect=yes. You can specify how many busy tones to wait before hanging up. The default is 3, but better results can be achieved if set to 6 or even 8. The higher the number, the more time is needed to detect a disconnected channel, but the lower the probability mistaking some other sound as being a busy tone.
  • busycount=5

• echocancel: Disable or enable echo cancellation (default is 'yes'). It is recommended that you do not turn this off. You may specify echocancel as 'yes' (128 taps), 'no' (0 taps, disabled), or a preset number of taps which are one of 16, 32, 64, 128, or 256. Each tap is one sample from the data stream, so on a T1 this will be 1/8000 of a second. Accordingly the number of taps equate to a 2ms, 4ms, 8ms, 16ms or 32ms tail length. Beware that if you set echocancel to a different value, Asterisk will fall back to the default of 128 taps without warning.
  • echocancel=no
• echocancelwhenbridged: Enables or disables echo cancellation during a bridged TDM call. In principle, TDM bridged calls should not require echo cancellation, but often times audio performance is improved with this option enabled. Default: no.
  • echocancelwhenbridged=yes
• echotraining: In some cases, the echo canceller doesn't train quickly enough and there is echo at the beginning of the call which then quickly fades out. Enabling echo training will cause Asterisk to briefly mute the channel, send an impulse, and use the impulse response to pre-train the echo canceller so it can start out with a much closer idea of the actual echo. However, the characteristics of some trunks may change as the endpoints become connected and, if there is a considerable delay between the circuit being 'up' and the endpoints being finalised, the training impulse may measure the characteristics of the open trunk rather than the completed circuit. Accordingly you may either specify a value between 10ms and 4000ms to delay before starting the impulse response process or 'yes', which equates to 400ms. Default: undefined.
  • echotraining=no
• rxgain: Adjusts receive gain. This is the audio recieved by Asterisk from the device. E.g: in a phone connected to a FXS channel, this would control the audio that is sent from the phone to Asterisk. This can be used to raise or lower the incoming volume to compensate for hardware differences. You specify gain as a decimal number from -100 to 100 representing dB. 10 is significantly high. Change these options by only a few dB at a time. Default value: 0.0
  • rxgain=4.2
• txgain: Adjusts transmit gain. This is the audio transmitted by Asterisk to the device. E.g: in a phone connected to a FXS device this would control the audio that is heard in the handset. This can be used to raise or lower the outgoing volume to compensate for hardware differences. Takes the same type of argument as rxgain. Default: 0.0
  • txgain=-10.2

lokal 31: čudno se ponaša - kada zovem 212643 - kad uspostavi konekciju on čučne

 -- Executing [212643@demo:3] Dial("SIP/31-0820a740", "zap/g1/033212643|400|tT") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g1/033212643
    -- Zap/2-1 is proceeding passing it to SIP/31-0820a740
    -- Channel 0/2, span 1 got hangup request, cause 17
    -- Zap/2-1 is busy
    -- Hungup 'Zap/2-1'
  == Everyone is busy/congested at this time (1:1/0/0)
  == Auto fallthrough, channel 'SIP/31-0820a740' status is 'BUSY'
[Aug 29 10:29:31] WARNING[8926]: chan_zap.c:8784 pri_fixup_principle: Call specified, but not found?  <<<<<<<<<<<<<<<<<<<<<<<<

što se tiče poziva dugih brojeva, kada sam ukinuo switch komandu stvar je proradila ?!?

033xxxxx sada ne radi:

  _0. => {
                //Set(TIMEOUT(digit)=5);
                //Set(TIMEOUT(absolute)=15);
                //Set(TIMEOUT(response)=60);

                &set_caller_id();

                switch (${EXTEN:0:5}) {
                        case 00492:
                                Dial(${SKYPE_TRUNK}/${EXTEN},400,tT);
                        default:
                                Dial(${TRUNK}/${EXTEN:${TRUNKMSDDIRECT}}, 400, tT);
                };

         };

033xxxxx sada radi

  _0. => {
                //Set(TIMEOUT(digit)=5);
                //Set(TIMEOUT(absolute)=15);
                //Set(TIMEOUT(response)=60);

                &set_caller_id();

                Dial(${TRUNK}/${EXTEN:${TRUNKMSDDIRECT}}, 400, tT);

         };

stavio na sip/31

dtmfmode=rfc2833

na telefonu stoji SIPINFO, Force RFC2833 Out-of-Band DTMF = yes

ne radi uopšte 217956