Hi everybody,
I had an ISDN card with winbond chipset and isdn4linux, but there was a lot of 
echo when call from SIP to ISDN, so I buy an HFC chipset card (Conceptronic 
c128i). I downloaded and compiled bristuff-0.2.0-RC5 and everything is going 
fine. The sound quality is excellent, there is no echo, but i have a strange 
problem. Sometimes calls made from SIP phones to ISDN just finished (asterisk 
drop the call). I was reading on this list and I readed about busydetect=no 
in zapata.conf (because in ISDN lines there is a busy signal in D-channel) 
but this don't help me. After that occurs I can dial any number at outside or 
receive calls. The only solution is to stop asterisk, unload zaphfc module, 
load again zaphfc module, run ztcfg, and run asterisk again. I'm using 
SJPhone as SIP softphones and two Grandstream (budgetone and handytone) 
hardphones.
Please help me because I cand find any solution about this. Here you have my 
config files:

[zapata.conf]
[channels]
language=es
switchtype = euroisdn
signalling = bri_cpe_ptmp
echocancel=yes
immediate=yes
group = 1
musiconhold=default
usecallerid=yes
callerid=asreceived
relaxdmtf=yes
busydetect=no
callprogress=no

overlapdial=yes

context=default
channel => 1-2

[zaptel.conf]
loadzone=es
defaultzone=es

span=1,1,3,ccs,ami
bchan=1-2
dchan=3

[indications.conf]
[es]
description = Spain
ringcadence = 1500,3000
dial = 425
busy = 425/200,0/200
ring = 425/1500,0/3000
congestion = 425/200,0/200,425/200,0/200,425/200,0/600
callwaiting = 425/175,0/175,425/175,0/3500
dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425
record = 1400/500,0/15000
info = 950/330,0/1000
dialout = 500

[sip.conf]
[a]
type=friend    
username=a
fromuser=a
callerid= A
host=dynamic            
nat=yes                       
canreinvite=no          
dtmfmode=rfc2833  
callgroup=1
pickupgroup=1


[extensions.conf]
TRUNK=Zap/g1               ;Zapata
exten => _9NXXXXXXXX,1,SetCallerID(9xxxxxxxx)  <-- (there is a real number)
exten => _9NXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}||r)
exten => _9NXXXXXXXX,3,Playtones(busy)
exten => _9NXXXXXXXX,4,Busy

The card don't share IRQ:
cat /proc/interrupts
          CPU0
  0:   49614493    IO-APIC-edge  timer
  1:        134    IO-APIC-edge  keyboard
  2:          0          XT-PIC  cascade
  8:          1    IO-APIC-edge  rtc
 14:     644986    IO-APIC-edge  ide0
 17: 3957387891   IO-APIC-level  zaphfc
 18:    8507911   IO-APIC-level  SiS 7012
 19:    9441882   IO-APIC-level  eth0
 20:          0   IO-APIC-level  acpi, usb-ohci
 21:         99   IO-APIC-level  usb-ohci
 22:          0   IO-APIC-level  usb-ohci
 23:          0   IO-APIC-level  ehci_hcd
NMI:          0
LOC:   49613413
ERR:          0
MIS:          0



And here are the messages from /var/log/asterisk/full:
Jan 26 08:45:03 VERBOSE[4481]:   == Primary D-Channel on span 1 up
Jan 26 08:45:04 DEBUG[4475]: Auto destroying call 
'[EMAIL PROTECTED]'
Jan 26 08:45:05 WARNING[4481]: PRI: !! Got a UA, but i'm in state 1
Jan 26 08:45:14 DEBUG[4475]: Setting NAT on RTP to 4
Jan 26 08:45:14 DEBUG[4475]: Check for res for a
Jan 26 08:45:14 DEBUG[4475]: Call from user 'a' is 1 out of 0
Jan 26 08:45:14 DEBUG[4475]: build_route: Contact hop: 
<sip:[EMAIL PROTECTED]:5060>
Jan 26 08:45:14 VERBOSE[5033]:     -- Executing 
SetCallerID("SIP/acasanova-cc59", "9XXXXXXXX") in new stack
Jan 26 08:45:14 VERBOSE[5033]:     -- Executing Dial("SIP/a-cc59", 
"Zap/g1/99999999||tT") in new stack
Jan 26 08:45:14 DEBUG[5033]: SIMPLE DIAL (NO URL)
Jan 26 08:45:14 VERBOSE[5033]:     -- Called g1/999999999
Jan 26 08:45:15 DEBUG[5033]: Ooh, format changed from unknown to ulaw
Jan 26 08:45:15 DEBUG[5033]: RTP NAT: Using address 192.168.1.212:16392
Jan 26 08:45:17 DEBUG[4481]: Queuing frame from PRI_EVENT_PROCEEDING on 
channel 0/1 span 1
Jan 26 08:45:17 VERBOSE[5033]:     -- Zap/1-1 is making progress passing it to 
SIP/acasanova-cc59
Jan 26 08:45:17 DEBUG[5033]: Dunno what to do with control type 15
Jan 26 08:45:19 DEBUG[4481]: Enabled echo cancellation on channel 1
Jan 26 08:45:19 VERBOSE[5033]:     -- Zap/1-1 is ringing
Jan 26 08:45:19 DEBUG[5033]: Driver for channel 'SIP/a-cc59' does not support 
indication 3, emulating it
Jan 26 08:45:19 DEBUG[5033]: Prodding channel 'SIP/a-cc59'
Jan 26 08:45:19 DEBUG[5033]: Scheduling timer at 160 sample intervals
Jan 26 08:45:19 DEBUG[5033]: Scheduling timer at 0 sample intervals
Jan 26 08:45:19 DEBUG[4475]: Auto destroying call 
'[EMAIL PROTECTED]'
Jan 26 08:45:28 VERBOSE[4481]: !! Unknown IE 124 (cs5, Unknown Information 
Element)
Jan 26 08:45:28 DEBUG[4481]: Echo cancellation already on
Jan 26 08:45:28 DEBUG[5033]: Dropping duplicate answer!
Jan 26 08:45:28 VERBOSE[5033]: !! Unknown IE 124 (cs5, Unknown Information 
Element)
    -- Zap/1-1 answered SIP/acasanova-cc59
Jan 26 08:45:28 DEBUG[4475]: Stopping retransmission on 
'[EMAIL PROTECTED]' of Respons
e 1: Found
Jan 26 08:45:37 VERBOSE[4481]:   == Primary D-Channel on span 1 down
Jan 26 08:45:37 DEBUG[5033]: Bridge stops because we're zombie or need a soft 
hangup: c0=SIP/acasanova-cc59, c1=Zap/1-1
, flags: No,No,No,Yes
Jan 26 08:45:37 DEBUG[5033]: Bridge stops bridging channels SIP/a-cc59 and 
Zap/1-1
Jan 26 08:45:37 DEBUG[5033]: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Jan 26 08:45:37 DEBUG[5033]: Hangup: channel: 1 index = 0, normal = 22, 
callwait = -1, thirdcall = -1Jan 26 08:45:37 DEBUG[5033]: disabled echo 
cancellation on channel 1
Jan 26 08:45:37 DEBUG[5033]: Set option TDD MODE, value: OFF(0) on Zap/1-1Jan 
26 08:45:37 DEBUG[5033]: Updated conferencing on 1, with 0 conference users
Jan 26 08:45:37 DEBUG[5033]: Set option AUDIO MODE, value: OFF(0) on 
Zap/1-1Jan 26 08:45:37 DEBUG[5033]: disabled echo cancellation on channel 1
Jan 26 08:45:37 VERBOSE[5033]:     -- Hungup 'Zap/1-1'Jan 26 08:45:37 
DEBUG[5033]: Exiting with DIALSTATUS=ANSWER.
Jan 26 08:45:37 VERBOSE[5033]:   == Spawn extension (default, 9999999999, 2) 
exited non-zero on 'SIP/a-cc59'
Jan 26 08:45:37 DEBUG[5033]: update_user_counter(acasanova) - decrement inUse 
counter
Jan 26 08:45:37 DEBUG[4475]: Stopping retransmission on 
'[EMAIL PROTECTED]' of Request 102: Found
Jan 26 08:45:37 WARNING[4481]: PRI: !! Got S-frame while link downJan 26 
08:45:37 WARNING[4481]: PRI: !! Got S-frame while link down
Jan 26 08:45:38 WARNING[4481]: PRI: !! Got S-frame while link downJan 26 
08:45:41 VERBOSE[4481]:   == Primary D-Channel on span 1 up
Jan 26 08:45:42 DEBUG[4475]: Auto destroying call 
'[EMAIL PROTECTED]'    


Thank you

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to