Is there any way to know if a call was transferred from reading the
CDR? Any relation in fields like UNIQUEID? Something that can be
scripted to make a special report?

Roger Burton West wrote:

> I want to hook one of them to the PSTN. Given that I am in
> the UK, what is a reasonably easily-available device to
> provide an FXO interface from a Linux box, with a minimum of
> faffing around with drivers? Just one line is needed, though
> in theory two might eventually be useful. My usual white-box
> hardware suppliers don’t seem to play in this field.

I’ve had good experiences with an OpenVox A400P, once you’ve done the “Dahdi
dance”, it settles down to be very reliable. Reasonable price, too. I bought
mine … Continue Reading

We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI)
cards.

I am having trouble completing faxes. Carrier send calls to me using SIP.
Any recommendation to have some success with Fax.

We trying using T.38 pass through and using G711U codec.

Asterisk Version 1.6.1.1

Thanks,

Dave

The Asterisk Development Team is pleased to announce the release of
DAHDI-Linux and DAHDI-Tools version 2.4.0.

DAHDI-Linux 2.4.0, DAHDI-Tools 2.4.0, and DAHDI-Linux-Complete are
available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

In addition to several bug fixes, the most significant changes from the
2.3.0 release are:

General DAHDI Changes:

* Added DAHDI_MAINT_ALARM_SIM maintenance mode for drivers that
support alarm simulation (wct4xxp). This is only used by
dahdi_maint and doesn’t change the ABI.

* Span callbacks are moved out of the dahdi_span structure potentially
saving memory when a single driver implements multiple spans.

Updated Drivers:

* wctdm24xxp, wcte12xp: Fix bug when … Continue Reading

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 … Continue Reading

Can i run asterisk on a openvz vps or do i need a kernel?
I dont use dadi

I have a pair of Asterisk servers which are happily routeing VoIP calls.
I want to hook one of them to the PSTN. Given that I am in the UK, what
is a reasonably easily-available device to provide an FXO interface from
a Linux box, with a minimum of faffing around with drivers? Just one
line is needed, though in theory two might eventually be useful. My
usual white-box hardware suppliers don’t seem to play in this field.

Also: I’ve heard good things about the PAP2T for getting analogue
handsets to talk to a VoIP server. But all the ones I … Continue Reading

Hi All,

In my dialplan and standard asterisk CLI logging i see that i am able to restrict the callerid when dialing out with asterisk.
however, on the receiving phone, the callerid is still displayed.

When i increment the logging of the pri with “pri set debug on span 1″ on the CLI i also get the lower level debugging info from the pri.
From here it looks like the SET CALLERPRES()=prohib is not working as expected… we see: “Presentation: Presentation permitted”

We use asterisk 1.6 with DAHDI and a PRI ISDN 30 line in the Netherlands.

Can anyone help me sorting … Continue Reading

Has anyone successfully made this scenario work in 1.4. I found info at
http://www.voip-info.org/wiki/view/Asterisk+presence indicating that this
does not work with 1.4 implementations.

We have a server that has been in operation since December of last year.
Two days ago we started seeing this messages over and over (maybe a couple
thousand in a minute):

[Sep 2 17:46:19] DEBUG[7422] audiohook.c: Write factory 0x2aaad40a0038 was
pretty quick last time, waiting for them.
[Sep 2 17:46:19] DEBUG[7421] chan_dahdi.c: Write returned -1 (Resource
temporarily unavailable) on channel 34

For the past two days Asterisk seems to fail at random intervals. It
does not crash but it stops processing calls. You need to restart Asterisk to
restore service. We are running Asterisk 1.6.2.11 with … Continue Reading

How does asterisk process URI’s that get sent to it?
I am having a issue with a Cisco phone, where 99% works except the call
forwarding. The phone issues a X-cisco-serviceuri-cfwdall which can be seen
when running a sip debug on the peer directly. However the system tries to
lookup the request as a extension, aka
X-cisco-serviceuri-cfwdall-
And of course can’t find this extension.

Is there any documentation regarding this out there specifically? I am
interested in how those references work, not just for the cfwdall, but other
URI’s as well.. The older cisco phones used to deal with Call forwarding … Continue Reading

There´s a way to get the channel signalling in dialplan?

I have changed the code in channels/chan_dahdi.c and includes:

< } else if (!strcasecmp(data, "signalling")) {

< ast_mutex_lock(&p->lock);

< snprintf(buf, len, "%s", sig2str(p->sig));

< ... Continue Reading

Hi,

I have a problem that the machine running asterisk 1.6.2.11 freezes
unexpectly time to time. Sometimes it runs for 4 weeks without any
problem, sometimes after a free it freezes again in 24 hours. But
usually it runs normally for 1 month or so before it freezes again.

I could not find any additional info in any file located in /var/log/
including asterisk’s messages file. Verbosity was set to 1000 but
nothing can be found which would indicate explain/indicate the problem.

If we connect a keyboard and a monitor to the server we see that the
machine is completely frozen, no login … Continue Reading

Hi everybody,

sometimes we have an Asterisk-crash, but no clue why this is happening,
so I’m trying to make a coredump to analyse it.

I compiled Asterisk 1.4.20.1 on CentOS 5.4 i386 with “DEBUG_THREADS” and
“DONT_OPTIMIZE”, then I start it with:
# /bin/bash /usr/sbin/safe_asterisk

This should do an “ulimit -c unlimited”, but I entered it in the
terminal again.

A
# ps -ef | grep asterisk
tells me that Asterisk is running as root and with the g-option for
writing a coredump:

root 21622 1 0 20:15 pts/0 00:00:00 /bin/bash
/usr/sbin/safe_asterisk
root … Continue Reading

Hello community,

I need to finish an AGI script when it invokes a macro from dialplan, how
can i do that? it’s quite confusing…the macro is making a hangup but the
script continues….

Thanks

Want to thank everyone who mailed; a couple of your ideas got me going
down certain paths, and found the answer here:

http://www.voip-info.org/wiki/view/Asterisk+tips+findme

Again, thanks!

-Ken

I’d *really* like to be able to have a call ring three different cell
phones; then, if someone answers, they have to somehow acknowledge the
call for it to be directed to them. That way, if one of the phones is
off, or out of range, it doesn’t go straight to that phone’s voicemail.

Asterisk 1.4 — though I could probably upgrade.

Suggestions on how to make this happen?

Thanks!

-Ken

1.4 will talk to 1.6 but results haven’t been as good in my experience as
native 1.4-1.4 or 1.6-1.6 communication.

Hi all

Been looking to find a way to stop the dtmf keys * 0 and # managing call
flow in the dialplan – I just want VM to stop recording on silence or
hangup.

I know I can trap the exit and loop back around but just want to ignore the
keys totally.

Any suggestions

P

Hi,

1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over.

2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I’ve added wW to the Dial() command, … Continue Reading