Podrška #15210
Zatvorenifold vzaphfc - all circuits are busy ? switch extensions.ael ? 31 ne može zvati fiksnu liniju
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root@ifold:~# cat /proc/zaptel/1
Span 1: ZTHFC1 "HFC-S PCI A Zaptel Driver card 0 [TE]" (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use)
Povezani tiketi 2 (0 otvoreno — 2 zatvorenih)
Izmjenjeno od Ernad Husremović prije skoro 17 godina
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
Izmjenjeno od Ernad Husremović prije skoro 17 godina
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
Izmjenjeno od Ernad Husremović prije skoro 17 godina
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.
Izmjenjeno od Ernad Husremović prije skoro 17 godina
dodao sam u zapata.conf
priindication=outofband
ali sada asterisk uopšte ne podiže zap interfejsa ?!
idem restartovati ifold
Izmjenjeno od Ernad Husremović prije skoro 17 godina
- uspio pozvati fiksnu liniju: 212643
- nisam uspio pozvati mobitel
- asterisk prima pozive moj_mobitel => officesa
Izmjenjeno od Ernad Husremović prije skoro 17 godina
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'
Izmjenjeno od Ernad Husremović prije skoro 17 godina
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
Izmjenjeno od Ernad Husremović prije skoro 17 godina
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
Izmjenjeno od Ernad Husremović prije skoro 17 godina
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? <<<<<<<<<<<<<<<<<<<<<<<<
Izmjenjeno od Ernad Husremović prije skoro 17 godina
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); };
Izmjenjeno od Ernad Husremović prije skoro 17 godina
- Naslov promijenjeno iz ifold vzaphfc - allcircuits are busy u ifold vzaphfc - all circuits are busy ? switch extensions.ael ? 31 ne može zvati fiksnu liniju
ovaj telefon 31 je pravo čudan (aastra 9133i)
kada sam u zapata stavio
busydetect=no ;busydetect=yes ;busycount=6
on je na kratko uspostavio poziv
-- Executing [217956@demo:3] Dial("SIP/31-b6303c28", "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/31-b6303c28 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/31-b6303c28
i odmah prekinuo vezu ?!
a konekcija sa mobitelom je ok
Izmjenjeno od Ernad Husremović prije skoro 17 godina
stavio na sip/31
dtmfmode=rfc2833
na telefonu stoji SIPINFO, Force RFC2833 Out-of-Band DTMF = yes
ne radi uopšte 217956
Izmjenjeno od Ernad Husremović prije skoro 17 godina
- Status promijenjeno iz Dodijeljeno u Zatvoreno
krajnji ishod: ifold je na hardy openvz kernel-u, asterisk ide preko zaptela, imao sam samo jedno zaglavljenje izlaznih poziva u ovom periodu, nakon restarta asteriska sve je proradilo
kvaliteta zvuka je odlična