Hi Guys,
Wondering if anyone has tried to make a direct SIP peer to peer call using
x-lite without any registrations of any sort. I can't seem to find the
setting.
Thanks,
bruce
--
_
-- Bandwidth and Colocation Provided by
Hi Guys,
I want to test my first video transmission call from Asterisk 1.6 to X-lite
softphone. I set videosupport=yes in SIP [general] and I have place a .wmv
file in /var/lib/asterisk/sounds/en and did a chown asterisk.asterisk on it.
I guess I have to use Playback command for the file and
send
picture back to me.
I have videosupport=yes in sip.conf [general] and I have allow=h263 in
sip.conf
How can I go about debugging the video transmission?
Thanks
On Sat, Apr 17, 2010 at 1:07 PM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:
On Sat, Apr 17, 2010 at 12:31 PM, bruce bruce
Thanks for the input. Problem was solved by adding transfer=no in
zapata.conf
For those who need TBCT, then add transfer=yes and facilityenable=yes in
zapata.conf.
However, if your telco has RLT or TBCT as a value added service and you have
not subscribed to it then you will face my problem if
Hi Guys,
I have been checking logs and noticed this over the last night. Should I be
concerned? and where to look for further details?
Sample:
[2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/1
successfully restarted on span 1
[2010-04-13 04:31:27] VERBOSE[3844] logger.c: --
Speaking of all these attacks, are there any good web managed security
monitor tools for CentOS out there that can be installed on the system so
that it can give us a visual of let's multiple failed attempts against SSH
or HTTPd?
Something nice that is simple and doesn't eat a lot resources and
Thanks, I can sleep better now.
On Tue, Apr 13, 2010 at 10:02 AM, Doug Lytle supp...@drdos.info wrote:
bruce bruce wrote:
[2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/1
successfully restarted on span 1
It's a normal function:
*resetinterval*: sets the time
Cool. I am just looking over splunk. Isn't that enough by it's own? or is
OSSEC needed to give it raw data? I think these two will take quite some
time to understand. Anything simpler out there as well?
Thanks,
Bruce
On Tue, Apr 13, 2010 at 10:42 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Hi Guys,
Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2,
LibPRI 1.4.10.
Placing a call into PRI and then transfering that call out to another
number. Problem is that the call rings out but the moment the other party
pickups both legs of the call are disconnected give
Nelson tnel...@rockbochs.com wrote:
- bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2,
LibPRI 1.4.10.
...etc
I was going to respond with some very insightful and helpful information
but I'm not a PRI Guru. Sorry
First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
*Sent:* Monday, April 12, 2010 2:22 PM
*To:* Asterisk Users Mailing List - Non
connected?
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
*Sent:* Monday, April 12
Hi Guys,
I am sorry if my issue is not related to this but I think it is.
I have a PRI with Bell Canada and when I dial in and have the call
transfered to a context to dial out and then have those two channels
bridged, the call disconnects with cause 16 just exactly as Jay R. Ashworth
shows in
.
Thanks,
Bruce
On Mon, Apr 12, 2010 at 8:51 PM, bruce bruce bruceb...@gmail.com wrote:
It just hit me that you are talking about TBCT. I don't think I am doing
TBCT as I still want both channels to keep two lines of my PRI occupied. In
addition, I would be interested to know how TBCT is done
, 2010 at 10:10 PM, bruce bruce bruceb...@gmail.com wrote:
Futher check into the PRI debug I am seeing this which actually relates to
TBCT and AOC-E error in /usr/src/libpri/pri_facility.c:
Message type: FACILITY (98)
[1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 03
Problem resolved with setting transfer=no in zapata.conf.
On Mon, Apr 12, 2010 at 9:14 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
I am sorry if my issue is not related to this but I think it is.
I have a PRI with Bell Canada and when I dial in and have the call
transfered
There you go. This confirms that SIP signaling determines where the calls
should go. I would take their word with a grain of salt specially with their
whole support center our of India. No disrespect, but it is bad service
overall.
-Bruce
On Sat, Apr 10, 2010 at 6:32 PM, Joshua Colp
out* of india.
On Sun, Apr 11, 2010 at 2:26 AM, bruce bruce bruceb...@gmail.com wrote:
There you go. This confirms that SIP signaling determines where the calls
should go. I would take their word with a grain of salt specially with their
whole support center our of India. No disrespect
Hi Guys,
Has anyone experienced this? Can I have a PRI guru weigh in on this?
Thanks,
Bruce
On Sat, Apr 10, 2010 at 3:46 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
I am calling out 416-999- on Channel 1 of PRI and then calling
416-999- on Channel 2 of PRI. When the two
Hi Guys,
I am calling out 416-999- on Channel 1 of PRI and then calling
416-999- on Channel 2 of PRI. When the two channels are going to be ZAP
native bridged, both channels hangup and CLI show PRI cause (16).
Asterisk Verbose *(Channel 1 already connected to party)*:
-- Requested
Just a week ago, I have been in the same situation. Provider was changing
from Cisco gateways to I think Nextone and hence provided me many IPs.
I found out that the media IPs don't matter and just played around with my
NAT settings and all calls can go through just fine by using simply:
LOLthis is just what I was facing 4 days ago. Unfortunately, Asterisk
doesn't provide a software feature in Zaptel to do a BUSY. But people on the
list suggest that one should call the telephone company and ask them to busy
it.
If you have the resource and don't mind the bill of calling the
Oh, I see. I haven't done a lot of testing on this new IP since the change
of gateways happened but I did Canada calls and they go fine. However, this
exact provider lies down to their teeth when it comes to problems of call
quality and calls not routing. They never accept faults. They even have
I really like the idea. I will try to ask. I don't know if they will be able
to do that easily though. They ask a week or two for any changes to the hunt
programming.
Thanks,
Bruce
On Thu, Apr 8, 2010 at 3:29 PM, Edo edo.eku...@gmail.com wrote:
Hello.. maybe you can just have the telco do an
: [asterisk-users] URGENT - How to exclude one ZAP channel for
outgoin and incoming calls
bruce bruce wrote:
Can I simply put ; in zapata.conf like this to seclude the first zap
line from getting calls in or out?
I'm not familiar with FreePBX, but I'd say that's logical. Make the
change
disable of the line now as it nears 9:00 A.M. operation
time. I will try that later in the day. I am amazed there is not much
control to the lines in situations like this.
Thanks for the inputs.
On Thu, Apr 8, 2010 at 8:43 AM, Doug Lytle supp...@drdos.info wrote:
bruce bruce wrote
, Apr 8, 2010 at 9:04 AM, Jeff LaCoursiere j...@jeff.net wrote:
On Thu, 8 Apr 2010, bruce bruce wrote:
I am not sure if unplugging line from card would work as it's still in a
hunt and calls will keep coming through that number and won't fall over
to
next line unless there is a BUSY
wrote:
Doug Lytle wrote:
Jeff LaCoursiere wrote:
On Thu, 8 Apr 2010, bruce bruce wrote:
Nope - unplugging a line that is in a hunt will result in
Ring-No-Answer.
Ditto for previous advice to destroy the zap channel or to leave it out
of
Our telecom guy said, that when
Not really when you got call center people who deal with makeup goods :-)
and their manager can only break things. I can't trust them anywhere near
the server. Let alone me telling them which cable to short on the bix. I
would presist for Digium to come up with something that would allow soft
HahahahaI definitly agree with Steve.
On Wed, Apr 7, 2010 at 11:44 AM, Steve Totaro
stot...@first-notification.com wrote:
On Wed, Apr 7, 2010 at 11:26 AM, Jason Walker
jason.wal...@amgsrv.comwrote:
I am getting a bunch of Primary D-Channel on span 1 up but there was not
a down
Hi Guys,
Currently, I have a Sangoma A400 installed with 20 ZAP PSTN analogue lines.
The first line is giving me problems due to rain (probably coroded line). My
server using FreePBX dials out with g0 (group 0 which includes all 20 lines)
and it happens that the bad line is the very first line.
Thanks for the update Jason,
How do the upgrades work if v1.6.0 is already install and one wants to
upgrade to 1.6.2 (once it's available)?
yum upgrade asterisk*
???
Thanks
On Mon, Apr 5, 2010 at 11:37 AM, Jason Parker jpar...@digium.com wrote:
Pablo Ruiz wrote:
Hello,
Does anyone
Yes, so this works (maybe safer than read=all and write=all):
read = system,call,command,agent,user,*originate*
write = system,call,command,agent,user,*originate*
I wasted probably a week on this - thanks to no documentation back in the
days with v1.6.
-Bruce
On Mon, Apr 5, 2010 at 1:50 PM,
I would suggest you try this. It works:
http://a2billing2asterisk.googlepages.com
On Mon, Apr 5, 2010 at 5:51 PM, Daniel Abreu dlab...@gmail.com wrote:
Hi guys. I am facing this problem here, using a2billing. error: 'Access
denied for user 'a2billinguser'@'localhost' (using password: YES)' I
..
Where are those 1.6.1/2 rpm's you are talking about??
On Sat, Apr 3, 2010 at 2:28 PM, bruce bruce bruceb...@gmail.com wrote:
RPMs for CentOS already exist. Though, I agree with better
notification/documentation for these and the keeping up with the updates.
On Sat, Apr 3, 2010 at 8:14 AM
RPMs for CentOS already exist. Though, I agree with better
notification/documentation for these and the keeping up with the updates.
On Sat, Apr 3, 2010 at 8:14 AM, Pablo Ruiz pablo.r...@gmail.com wrote:
Hello,
Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary
packages
I think you have caller ID update set to Yes and A2Billing first asks you
to: Enter your Caller ID number and then it asks you: Enter your
destination number while you mistake both for destination number.
Otherwise, I am confused by the title of your question that your caller id
doesn't pass and
SugarCRM and the church. This sounds just like a business; one that doesn't
like to call itself a business but employees tactics. I suggest providing
them with a solid cisco system with 100s of thousands dollars in cost where
they will have less money left to do bad things to world. Asterisk is
Hi Everyone,
I have tried to set the box to DMZ and also tried to port forward 5060
TCP/UDP and 1/2 UDP to the server IP but it's no use and there is a
no audio issue. I am pretty certain it's a NAT issue as the sip call
establishes. I also made a succesful IAX2 call through IAX trunking
Hi Guys,
I have a need to alter the general timeout in Asterisk. I am wondering if
this is something that is hard coded into Asterisk code or if there is a
parameter that can be set somewhere.
For outbound, I am using x. and hence unless I append a # sign, I would have
to wait maybe 5 seconds or
Hi Everyone,
I have a provider who is asking me to send SIP signals through
111.111.111.111 and then media through Media 1: 222.222.22.222 and Media 2:
244.244.244.244. This provider authenticates by IP and I think is using
Sonus gear and hence they have some load balancer or something...
I have
to check your sip phone's dialout pattern and timeout values.
--
Zeeshan A Zakaria
On 2010-03-20 10:58 AM, Doug Lytle supp...@drdos.info wrote:
bruce bruce wrote:
For outbound, I am using x. and hence unless I append a # sign, I
would ha...
You really do need to give us a snippet
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