I have Asterisk 1.2.29, Zaptel 1.2.24 and Freepbx Setup for
a queue up to 15 agents through a PRI line, it was working fine for more than 1
year, suddenly, when there is a load on the queue, the asterisk service
disconnects and the calls are dropped. And the service starts again after few
Hi,
In our office, we're slowly migrating from a cisco call manager set up
to asterisk. Problem is management doesn't want to buy any other
hardware as they had already invested a lot in cisco. The main cause
of this is asterisk's added features like unique FAX number for
everyone in the company
On 16 May 2009, at 12:46, Timothy Smith wrote:
blah
Has anyone had the above set up working successfully? Attached are
some confs.
Thanks a lot for your assistance.
Check about the sip.conf 'insecure' option. I have had to use it in
the past for similar stuff. I think it was
Thanks Steve for this tip.
I have insecure=very is not yet deprecated. I have added it but still no good.
I personally think the problem could be with the codecs. Any ideas?
I have attached some debug info.
Regards,
Tim
On Sat, May 16, 2009 at 3:25 PM, Steve Howes st...@geekinter.net wrote:
sean darcy wrote:
sean darcy wrote:
Mark Michelson wrote:
sean darcy wrote:
Danny Nicholas wrote:
You lost conf-getconfno.gsm . Asterisk is trying to play that file to
let
you pick a conference number to use. It goes in /var/lib/asterisk/sounds.
Grep for it.
-Original Message-
On Sat, May 16, 2009 at 7:46 AM, Timothy Smith timotsm...@gmail.com wrote:
I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
and also a dialpeer to forward on the router to forward calls to my
asterisk. It works properly but the problem is there is NO AUDIO! I
have tried to
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
extensions.conf:
[meetme]
exten = 2663,1,MeetMe(,De)
exten = 2663,n,Hangup()
exten = 2666,1,MeetMe()
exten = 2666,n,Hangup()
What I'm expecting is to dial 2663, get a conference room number ( 600,
I suppose
Hi Carlos
Agentcallbacklogin was deprecated in Asterisk 1.4 and eliminated from
1.6 so you now need to use Dynamic Agents. Although they claim that is
is simple enough to replace that functionality with dial plan code I
have yet to see a one line example that replaces everything the
Hi Jim
Thanks for your code!! I see you use the Voicemail system to authenticate,
have you ever managed to avoid that as I don't use voicemail at all and I am
thinking if I use that solution I will need to set up a voicemail for all
the queue members just to get them to log in.
hehe What were
David Anthony O Reilly wrote:
hehe What were the developers thinking by removing the old system! It
worked perfect!! and by the looks of it nobody has ever recovered from
the command removal unless they hack around with the voicemail system.
I think the best solution is to either use an AGI
David,
Thanks a lot for your input. I will enable DSP farming. Like some
other techies, I just wanted to see it work before i consider others
things.
I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these
Where do you want to get the value that agent uses to validate?
You can do your own code to do the validation.
Get the value from where ever and then do a read and compare the value read
with the value you retrieved from where ever. If there is match you are
done if no match say error, maybe
On Saturday 16 May 2009 08:21:43 sean darcy wrote:
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
extensions.conf:
[meetme]
exten = 2663,1,MeetMe(,De)
exten = 2663,n,Hangup()
exten = 2666,1,MeetMe()
exten = 2666,n,Hangup()
What I'm expecting is to
Steve Howes schrieb:
Check about the sip.conf 'insecure' option. I have had to use it in
the past for similar stuff. I think it was 'insecure=very' but that
might be deprecated by now..
insecure=very should now be written as insecure=port,invite
Philipp Kempgen
--
AMOOMA GmbH -
Tilghman Lesher wrote:
On Saturday 16 May 2009 08:21:43 sean darcy wrote:
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
extensions.conf:
[meetme]
exten = 2663,1,MeetMe(,De)
exten = 2663,n,Hangup()
exten = 2666,1,MeetMe()
exten = 2666,n,Hangup()
I have put the following in my voicemail.conf-file :
mailcmd=/usr/local/bin/msmtp -d --syslog=on
-d and syslog=on are to debug some information, because I am still not
receiving my voicemail-messages via mail as an attachment !
I don't know which mailcommand I need to put here to make Asterisk
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