[asterisk-users] how to improve sound file quality?
We have recorded wav files with 44k, 22k, 16k, 11k and 8k Asterisk does not accept these wav files. I used sox input.wav output.gsm to get them to work. However, the only the 8k file did convert and the quality is poor. How can I improve the quality? bye Ronald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can asterisk work with a dynamic IP?
I know I can setup asterisk without Internet at all and it works as local pbx. Would an asterisk box work with a dynamic IP, with a dyndns name? What must I take care if I try that? bye Ronald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Solved] Wellgate & Asterisk
Guillermo Salas M. wrote: > El jue, 27-11-2008 a las 21:05 +0800, Ronald Wiplinger (Lists) escribió: > >> I got a Wellgate 3804A and need some hints: >> >> Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate >> >> Wellgate 3804A settings (Line1~Line4): >> > > > I've one wellgate 3804 (old version) with 4 fxo ports integrated with > asterisk 1.4. > > Regards, > > I could solve it! I had to add routing in the 3804A. Now both, dialin and dialout is working. bye Ronald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wellgate & Asterisk
I got a Wellgate 3804A and need some hints:
Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate
Wellgate 3804A settings (Line1~Line4):
1. Sip Config
Mode: Proxy
Primary Proxy IP Address: *.131
Primary Proxy port: 5060
Line1 Number: 1002
2. Security Config
Line1 Account: 1002
Line1 Password: **
3. Line Configuration
Line1: Type=FXO, Hunting Group=2, Hot Line = 88621002
Asterisk settings:
users.conf:
[1002]
context = DID_1002
host = *.133
username = 1002
secret = **
trunkname = WellGate-1002 ; GUI metadata
hasiax = no
registeriax = no
hassip = yes
registersip = yes
trunkstyle = voip
hasexten = no
host = dynamic
disallow = all
allow = ulaw,alaw,gsm,g726,g729
extensions.conf
1002 = SIP/1002
...
[DID_1002]
exten => _88621002,1,NoOp(${CALLERID(num)})
exten => _88621002,n,Wait(1)
exten => _88621002,n,SayUnixTime
include = DID_1001_timeinterval_working day|${timeinterval_working day}
include = DID_1001_default
[DID_1001_default]
exten => s,1,NoOp,${CALLERID(num)}-${CALLERID(name)}
exten => s,n,Answer
exten => s,n,zapateller(nocallerid) ; torture telemarketers
exten => s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,n,Hangup
include = default
[DID_1001_timeinterval_working day]
exten = _6888,1,Goto(default|6888|1)
If I call in at line2, then I can hear the Time announcement and I can
dial during that announcement an extension number.
BTW, where can I find the additional sounds I had at an previous setup
(If you know the extension, ...), which should replace the SayUnixTime
I have no idea how to get dial out to work. Can anybody give me a hint,
please?
In Asterisk I see:
[Nov 27 20:58:00] NOTICE[5095]: chan_sip.c:9227 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #102)
-- Got SIP response 486 "Busy Here" back from *.133
*CLI> sip show peers
1002/1002 *.133D 5060 Unmonitored
*CLI> sip show users
1002 **
DID_1002 No RFC3581
*CLI> sip show registry
*.133:5060 1002 120 Request Sent
bye
Ronald
___
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[asterisk-users] [SOLVED] Re: Upgrade 1.4.19 to 1.6 => segementation fault
Ronald Wiplinger (Lists) wrote: > During compiling I have not seen an error, however, when I start > asterisk again it ends with: > > > app_morsecode.so => (Morse code) > == Registered custom function 'SYSINFO' > func_sysinfo.so => (System information related functions) > Segmentation fault (core dumped) > > > How can I figure out what is wrong? > I removed all modules, which were left from the 1.4 installation and now it works! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrade 1.4.19 to 1.6 => segementation fault
During compiling I have not seen an error, however, when I start asterisk again it ends with: app_morsecode.so => (Morse code) == Registered custom function 'SYSINFO' func_sysinfo.so => (System information related functions) Segmentation fault (core dumped) How can I figure out what is wrong? bye Ronald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom - we are puzzled
we have installed asterisk and snom with PUBLIC IPs (IP/25) on one DSL line we have for our office a different ADSL with one IP shared. Two identical setup snom 360 (except the user name) with two public IP addresses are connected at the hub to the server / DSL line phone A can call B, B cannot call A, because A is not registered!!! We disconnect A and setup a softphone (on the ADSL line with stun) and it works. How can I track down this problem. bye R. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
