Displaying 10 results from an estimated 10 matches for "makecall".
Did you mean:
makeall
2004 Apr 03
0
Grandstream and codec G.711
...ack
> >
> > First, this should be:
> > Dial(OH323/483317839|20|tT) not
> > Dial(OH323/483317839|20|t|T)
> >
> > > [2]WrapperAPI::h323_make_call: Making call.
> > > [2]WrapMutex::Wait: Requesting mutex callMutex
> [wrapendpoint.cxx, 269,
> > MakeCall]
> > > [2]WrapMutex::Wait: Got mutex callMutex
> [wrapendpoint.cxx, 269, MakeCall]
> > > [2]WrapH323EndPoint::MakeCall: Making call to
> 483317839
> > > [3]WrapH323Connection::WrapH323Connection:
Outgoing
> capability
> > > G.711-uLaw-64k{hw}
> > &...
2015 Mar 30
0
WaitForSilence NEVER detects silence,,Post
...7.0.0.1/playmessage,${CALLID});
> AGI(agi://127.0.0.1/saytext,"Goodbye.");
> Hangup();
> }
And the CLI just outputs:
> == Using SIP RTP CoS mark 5
> > Channel SIP/twilio-0000006e was answered
> -- Executing [100 at makeCall:1] Answer("SIP/twilio-0000006e", "") in
> new stack
> -- Executing [100 at makeCall:2]
> WaitForSilence("SIP/twilio-0000006e", "5000,2,60") in new stack
> -- Waiting 2 time(s) for 5000 ms silence with 60 timeout
> -- Exiting with...
2015 Mar 30
0
WaitForSilence NEVER detects silence
...7.0.0.1/playmessage,${CALLID});
> AGI(agi://127.0.0.1/saytext,"Goodbye.");
> Hangup();
> }
And the CLI just outputs:
> == Using SIP RTP CoS mark 5
> > Channel SIP/twilio-0000006e was answered
> -- Executing [100 at makeCall:1] Answer("SIP/twilio-0000006e", "") in
> new stack
> -- Executing [100 at makeCall:2]
> WaitForSilence("SIP/twilio-0000006e", "5000,2,60") in new stack
> -- Waiting 2 time(s) for 5000 ms silence with 60 timeout
> -- Exiting with...
2007 Jun 26
1
Modification of Caller ID based on context
Hi,
I have been looking for an example of accomplishing this, but I've been
unable to locate something similar to what I'm trying to do.
Here's the scenario:
Users caller ID is set to their internal extension (200-250). This is set in
sip.conf for each user. Each user has a local DID as well (hosted through
Vitelity, for example (555)111-2222). The problem is that this extension was
2015 May 06
2
can ooh323 work with cisco router?
hello every body,
i have big problem to configure h323 trunk between cisco router and
asterisk 11.13.1 which uses ooh323 module. any body knows if ooh323 module
can work with cisco routers or not???? (in gateway mode, it is ok and
register in cisco gatekeeper but i can not configure trunk h323)
any comments or hints are really appreciated.
SAM
-------------- next part --------------
An HTML
2015 May 06
2
can ooh323 work with cisco router?
...100 to 200, everything is ok but when i call from 200 to
100, phone rings but after i answer it, i have no voice and call terminates
after 5 seconds. this is ooh323 debug(in asterisk11.13.1 system):
ooh323_get_rtp_peer OOH323/peer-2-5 -> (null):0, 1
this ia h322_log:
10:42:10:835 Processing MakeCall command ooh323c_o_3
10:42:10:835 Created a new call (outgoing, ooh323c_o_3)
10:42:10:835 Enabled RTP/CISCO DTMF capability for (outgoing, ooh323c_o_3)
10:42:10:835 Enabled RFC2833 DTMF capability for (outgoing, ooh323c_o_3)
10:42:10:835 Dtmf mode set to H.245(alphanumeric) for (outgoing,
ooh323...
2004 Apr 18
0
OpenPhone <-> Asterisk w/H.323
...ting,
I've gotten some intermittent results however. All my calls are from a PC
to asterisk - I don't have an outbound requirement.
If anyone has successfully made either of these combo's work, could you
please suggest some area where I may have gone wrong?
With OpenPhone:
When using MakeCall on OpenPhone, asterisk answers the call fine, even
though the OpenPhone display still shows "ringing" throughout the duration
of the call. The audio (G7.11 uLaw) comes through the PC speaker fine.
When I issue Hangup on OpenPhone, asterisk most of the time does not get a
hang up signal,...
2006 May 24
1
Placing call files in/var/spool/asterisk/outgoing/ does not work
> you should mv the file (and in the same filesystem, so 'rename' is
used)
>
You might want to chmod or even chown the file first as well. I wrote a
little script that does all of this before the .call file is mv'd into
the outgoing directory:
cp /tmp/test3.call /tmp/test1.call
chmod 666 /tmp/test1.call
chgrp asterisk /tmp/test1.call
chown asterisk /tmp/test1.call
mv
2005 May 13
3
2 minutes pause before ring on H323 channel
...inute exactly (60 seconds)
exten => 20,1,Dial(H323/h323phone) ; this leads to 120 seconds pause before ring
exten => 21,1,Dial(H323/h323phone@192.168.0.101) ; this leads to 60 seconds pause before ring
After quick debugging session I found that this time goes to the call to
H323EndPoint::MakeCallLocked(fullAddress, token, opts) in MyH323EndPoint::MakeCall function.
MakeCallLocked is part of OpenH323 and this is too deep for me... I'm not sure is this is a
problem of OpenH323 or of channel driver but my speculation is that the time goes in some kind of
timeout wait... who knows...
Does...
2019 Jan 09
2
Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)
...t;h,n(done),noOp(Call Completed)
and the VCCALLOUT macro:
[macro-VCCALLOUT] ;macro to dial numbers ; ${ARG1} Channel To Use ; ${ARG2} Number To Dial ; ${ARG3} FailOver Channel ; ${ARG4} FailOverNumber
exten=>s,n(setchan),Set(chantouse=${ARG1})
exten=>s,n,Set(numtodial=${ARG2})
exten=>s,n(makecall),GotoIf($["${timeLimit}" = ""]?dialNoLimit:dialLimit)
exten=>s,n(dialNoLimit),Dial(${chantouse}/${numtodial},60,TL(3900000:60000:30000))
exten=>s,n,NoOp(Dial Status: ${DIALSTATUS})
exten=>s,n,GoTo(s-${DIALSTATUS},1)
exten=>s,n(dialLimit),Dial(${chantouse}/${numtodial...