Hi,

(I have install trixbox2.8 with asterisk 1.6)
I am using speex codec for my Inter asterisk communication

Question1: I want to configure speex on 2.15kbs and packetization of 60ms seconds for that is have configured “codecs.conf” for desired result and also placed a line in general section of “sip.conf” allow=speex:60 after disallow=all line .

I have also configure SIP trunk between two asterisk to use speex:60
During debugging I have checked that both side accept speex as a codec for call and ptime:60 but

I am facing following unexpected results

1-> When I check the packet rate from one asterisk to other asterisk for one call its not (1000/60 == 17)?

2-> When ever I change the softphone result changes i.e. data ratae chages ?

3-> How can I use my own codec “xyz” in asterisk to place calls ?

4->if I change the codecs.conf then no results appears in packet size which is comming out of asterisk?

Hello List,

Maybe I’m mistaken, but, shouldn’t the ‘fname_base’ variable of
‘Monitor’ application affect the file name generated through ‘automon’
feature?

I initialized this variable with a value as follows:
Set(fname_base=auto-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})

a. Should I use ‘fname_base’ in uppercase (FNAME_BASE)? or…
b. Is this variable independent of the ‘automon’ feature?

Thanks in advice,

PS.
version: Asterisk 1.4.33.1
OS: Slackware Linux 13.0

Hi,

It seems ive broken my settings and now, asterisk isnt detecting my DTMF tones.

What kind of diagnostics can I do to work this out?

I’ve set the extension in sip.conf to everything listed on this page but no result. I’ve also played around with the settings on the phone with no help either. Someone once said on here that Asterisk and the SIP phone have to match, but that doesnt seem to work either.

Any ideas?

Thanks
Dan

I use Asterisk 1.6.2.11 and this is my dialplan:

[test]
exten => 9999,1,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten => 9999,n,Answer()
exten => 9999,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten => 9999,n,PlayBack(hello-world)
exten => 9999,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten => 9999,n,MYSQL(Set timeout 2)
exten => 9999,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten => 9999,n,MYSQL(Connect connid localhost user pass asterisk)
exten => 9999,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten => 9999,n,MYSQL(Query resultid ${connid} SELECT SLEEP(10))
exten => 9999,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten => 9999,n,MYSQL(Fetch fetchid ${resultid} RESULT)
exten => 9999,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten => 9999,n,MYSQL(Clear ${resultid})
exten => 9999,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten => 9999,n,MYSQL(Disconnect ${connid})
exten => 9999,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten => 9999,n,NoOp(Result: ${RESULT})
exten => 9999,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)})
exten => 9999,n,Hangup()

When i call to 9999 this is the CLI output:

Connected to Asterisk 1.6.2.11 currently running on Asterisk (pid = 2092)
Verbosity is at least 2147483647
Asterisk*CLI>
== Using SIP RTP CoS mark 5

Hello,

When an agent does a blind transfer the call hangups for him but shows as
“In use” in queue in my CRM (used for auto dialing). As a result the agent
have to wait until the transfered call completes. Is there any way to change
this behaviour ?