Hi Douglas,

You;re right, that method is useful only for one-to-one call but as soon as
the call gets transferred etc etc as you mentioned everything will get
mixed and confusing.

Any way I this can be done? Can’t a call be passed off from one channel to
> another, which would leave me with only seeing a part of the logs for the
> life of the call if I only grep the logs based on one channel id?

Yes, is the answer if you want this to implement. You need to do the
following in order to achieve an start-end logging of a call.

1- As soon as call Enters Asterisk dialplan save its UNIQUEID (plus any
other key i.e timestamp ) in a CALL-IDENTIFY variable.
2- Use the CALL-IDENTIFY variable throughout your dialplan contexts to
verbose() useful information.(I saw a log() application in asterisk 1.8.5
to do this in a the log file…i.e print logs of your own)

Another interesting thing for this purpose would be CEL, though it maynot
be available in your older deployments. I haven’t toyed around with CEL
myself but so far I’ve the impression that its a very verbose form of CDRs.
So using CEL to keep track of your call in a DB would help as well.

Another Idea is to use the SIP-Header Call-ID as your CALL-IDENTIFY
variable. This way when you’re debugging the issue using asterisk logs
alongside taking SIP-traces it’ll help you identify which packets belongs
to which log lines.

Wireshark is a great tool. I take Sip traces, open up in wireshark goto
voip calls and you’ll see all the calls that were at-least initiated after
when u started the trace. Apply filter on your specific call and see only
sip traces relevant to one particular call.

Thats all I could come up at this time.

I hope this would be of some help.

On Sat, Oct 29, 2011 at 3:14 PM, Eric van der Vlist wrote:
> Hi,
>
> Xorcom astribanks get initialized straight on when using Ubuntu 11.10
> packages but I am having a hard time to get the same result running in a
> qemu/libvirt image.
>
> The first difficulty is that astribanks devices get different usb device
> ids during their initialisation process, requiring hot plug support.
>
> I have figured out how to solve this issue using the technique described
> in this post :
> http://www.blogs.uni-osnabrueck.de/rotapken/2011/04/11/how-to-auto-hotplug-usb-devices-to-libvirt-vms/
>
> That doesn’t seem to be enough and the initialisation fails with a
> status 1:
>
> Oct 28 18:58:19 asterisk-rg ‘xpp_fxloader’[1006]: Trying to find what to
> do for product e4e4/1160/101, device /dev/bus/usb/001/004
> Oct 28 18:58:19 asterisk-rg ‘xpp_fxloader’[1010]: Loading firmware
> ‘/usr/share/dahdi/USB_FW.hex’ into ‘/dev/bus/usb/001/004′
> Oct 28 18:58:23 asterisk-rg ‘xpp_fxloader’[1024]: Trying to find what to
> do for product e4e4/1161/101, device /dev/bus/usb/001/005
> Oct 28 18:58:34 asterisk-rg
> ‘xpp_fxloader’[1035]: /usr/sbin/astribank_tool failed with status 1
>
> Seeing that Xorcom requires USB 2.0 and that the current versions of
> libvirt and qemu in Ubuntu 11.10 emulate USB 1.10 in guests, I have
> installed Boris Derzhavets’ packages:
> https://launchpad.net/~bderzhavets/+archive/seabios163 and updated my
> host definition to emulate USB 2.0 but I still have the same issue.
>
> Have I missed something?
>
> Thanks,
>
> Eric
>

Try Xorcom’s great support…. Tzafrir posted the solution to this a
few months ago, search the list for it.

On Saturday 29 Oct 2011, Raj Mathur (राज माथुर) wrote:
> [snip]
> Callers coming in from the PSTN (through the Dial server, over IAX2)
> can also talk normally after an agent has picked up the call.
> However, callers from the PSTN get the announcement and/or MOH
> blanked out after a random period of time, typically 5-10 seconds.
> This often happens in the middle of the queue position or thank-you
> announcement.
>
> After the blanking out, the call is still alive, queue functions are
> working, and if an agent picks up the calls s/he can talk normally to
> the caller. However, blanking out of the MOH/announcement makes the
> caller think that the call has been dropped, and they hang up before
> an agent answers.
>
> Debug logs show that Asterisk is playing the MOH and announcement
> files continuously even though the caller cannot hear them.
>
> Unable to figure out why the blanking happens ONLY on incoming calls
> from the PSTN. Any help appreciated.

Further simplified the issue to an extension that just does:

… Answer()
… MusicOnHold(default)

When called from the PSTN, the musiconhold blanks out after a few
seconds, while it plays fine when the extension is called locally.

Regards,

using asterisk-10 on CentOS

I am trying to get googleapps calendar integrated with my system.
However, following all the instructions that I can find it still
fails. this is my config file:

[myGoogleCal]
type=caldav
url=https://www.google.com/calendar/dav/< >/events/
user=< >
secret=< >
refresh=15
timeframe=60

when I start asterisk, and type “calendar show calendars” I get

genesis2*CLI> calendar show calendars
Calendar Type Status

I just had a weird experience. My Asterisk installation stopped
working, and upon investigation I found that the ownership of several
files had changed from asterisk:asterisk to root:root. The files in
question were:

/etc/asterisk/extensions.conf
/etc/asterisk/features.conf
/etc/asterisk/rtp.conf
/etc/asterisk/sip.conf
/etc/asterisk/voicemail.conf
/var/log/asterisk/cdr-csv/Master.csv
/var/spool/asterisk/astdb

Asterisk has been running for quite a while (several weeks at least)
since I last modified any of the configuration files.

I did reboot earlier today when I was unable to cleanly stop BIND on
this box. (32-bit Scientific Linux 6.1, BTW.)

Anyone have any idea what just happened?

TIA!