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
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
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
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
..
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
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
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.
: [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
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
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
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,
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
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,
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
, Apr 19, 2010 at 12:08 AM, Alyed al...@vivoxie.com wrote:
You can't do that with Xlite, try Sjphone instead.
Alyed
2010/4/17 bruce bruce bruceb...@gmail.com
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
Hello Everyone,
I have a system that was working on Sunday 1 P.M. and then gives Congestion
on Monday morning. Sometimes over night it probably stopped working. It's a
PBXinaFLASH with Asterisk 1.4 and libPRI with a 23 channel PRI connected and
24th D-Channel.
This is all I see in
I've never been able to with xlite
it's just with Sjphone it's straight forward.
Alyed
2010/4/19 bruce bruce bruceb...@gmail.com
That is not correct. It's possible by adding a display name and adding the
IP address of the pbx you are calling as the host ip. Then uncheck the
register button
Dial: 0
Logical Channel Mapping: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T309 Timer: -1
T313 Timer: 4000
N200 Counter: 3
thanks,
Bruce
On Mon, Apr 19, 2010 at 3:00 PM, Doug Lytle supp...@drdos.info wrote:
bruce bruce wrote:
[2010-04-19 08:45:50] WARNING
I have had problems with Portech firmware using Chrome browser. The problem
was that when I changed the password on the gateway it would apply that
password to SIP PEERS as well. So, maybe, you are actually not having the
right password in your SIP peer as well and hence your Asterisk sends
a charm with an IP-phone
(Grandstream) ?!
Jonas.
bruce bruce wrote:
I have had problems with Portech firmware using Chrome browser. The problem
was that when I changed the password on the gateway it would apply that
password to SIP PEERS as well. So, maybe, you are actually not having
Hi Everyone,
I have a weired situation where calls in and out are proceessed all right
but when I dial *97 Asterisk is literally choking when it comes to
announcements like Password or Call from 205-456-. Each one of those
announcements can take like 10+ seconds to finish with most of it not
Yes, it's all g.711 ulaw.
On Wed, Apr 21, 2010 at 1:37 PM, Darrick Hartman (lists)
dhart...@djhsolutions.com wrote:
Are your sound files being transcoded or played back in their native
formats?
On 04/21/2010 12:25 PM, bruce bruce wrote:
Hi Everyone,
I have a weired situation where
yes, it's on Amazon.
On Wed, Apr 21, 2010 at 2:26 PM, Ryan Bullock rrb3...@gmail.com wrote:
Are you running asterisk in a virtual machine?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Thanks for the input.
I am going to check this once I get access to system again tonight.
But I thought the timing source dahdi_dummy is only good for features like
MeetMe or conference rooms? or am I wrong and it has an effect on any type
of calls and checking voice messages?
Thanks
On Wed,
tell from these?
On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the input.
I am going to check this once I get access to system again tonight.
But I thought the timing source dahdi_dummy is only good for features like
MeetMe or conference rooms? or am I
at 7:56 PM, Sean Brady sbr...@gtfservices.com wrote:
On 04/21/2010 05:36 PM, bruce bruce wrote:
Here are result of dahdi_test:
[r...@ip-10-251-123-3 ~]# dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434%
-434.763
I know that anything lower than 99% is bad. But *-400 *?
Anything care of comment?
Thanks,
On Wed, Apr 21, 2010 at 7:45 PM, Steve Howes steve-li...@geekinter.netwrote:
On 22 Apr 2010, at 00:36, bruce bruce wrote:
Opened pseudo dahdi interface, measuring accuracy...
99.725% 96.018% 99.532
at anytime on this server.
Thanks
On Wed, Apr 21, 2010 at 8:03 PM, Carlos Chavez cur...@telecomabmex.comwrote:
On Wed, 2010-04-21 at 19:36 -0400, bruce bruce wrote:
Here are result of dahdi_test:
[r...@ip-10-251-123-3 ~]# dahdi_test
Opened pseudo dahdi interface, measuring accuracy
WWW-Authenticate: Digest algorithm=MD5, realm=103001vc, nonce=03e68412
Content-Length: 0
Jonas.
bruce bruce wrote:
Try changing port=5064 to port=5060 in your Asterisk config file. Portech
will negotiate it's port with Asterisk itself
Take out the router/firewall and connect directly to the net to test your
NAT problem theory.
On Thu, Apr 22, 2010 at 12:15 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
Jared,
thank you for your answer.
As I said in my previous mail, I'm using a Zyxel NBG-419 router (which
normally
I have a list of CLIDs prefixes that I want to use in a context.
Basically, I want to do this but the list of prefix numbers is much longer.
List of prefixes (556,557,557,989.)
[custom-inbound]
exten = _556,1,answer
exten = _556,n,playback(beep)
exten = _557,1,answer
exten =
Adobe Air and Adobe FMS are good examples of VoIP working flawlessly over
TCP. We are actually developing a flash phone which needs only TCP to
transmit both signal and audio.
-Bruce
On Sat, Apr 24, 2010 at 2:01 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
RTP stands for Real-Time Transport
Hi Everyone,
How is this possible? How can I go about debugging this? I think that the
sound chopping and choking is also related to this. I have never seen
Asterisk show 43% of cpu usuage when there is only one call going. It
actually flactuates down to 11% and up to 43%.
Please guide me as to
with the
other party.
You are sending FACILITY - maybe the other party does not like FACILITY and
hangs up.
IIRC there is a setting in zapata.conf to enable/disable FACILITY.
regards
klaus
Am 10.04.2010 21:46, schrieb bruce bruce:
Hi Guys,
I am calling out 416-999- on Channel 1 of PRI
Hi Guys,
Anyone might know why this error keeps showing up and inbound/outbound is
not working on a Bell PRI with Sangoma A101D?
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement
No calls can be made inbound/outbound.
Keeps repeating. No alarms ON and no changes been made to
Hello Everyone,
Are these indications of attacks on this system? I specifically have port 22
disabled at all times and only port forward it to server when I access SSH
for a minute or so. Shouldn't UNKNOWN be an actual IP address?
*/var/log/secure:*
May 14 00:35:39 pbx sshd[9011]: Did not
Unplugging just turns off the phone and has no effect on the settings. You
can not damage the phone by tampering configurations but you can mess up
the settings and it might not register, send, or receive calls.
User manu for your reference:
Hi Guys,
We have a problem with Caller ID not being displayed. I want to test
everything to see where the problem is with the incoming Caller ID and why
it's not displaying.
I am tracking this down to Presentation prohibited of network provided
number even though the Caller doesn't use *67 and
Hi Guys,
Can q931.c be re-compiled using gcc or something else without the need to
re-do the whole libpri? Some changes were made to q931.c and I want those to
be reflected in .a .o .so .lo files as I think those are the files read by
Asterisk vs the .c file.
Thanks,
--
, May 15, 2010 at 4:56 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Sat, May 15, 2010 at 04:32:19PM -0400, bruce bruce wrote:
Hi Guys,
Can q931.c be re-compiled using gcc or something else without the need to
re-do the whole libpri? Some changes were made to q931.c and I want those
Maybe drop the call in a Meetme room and have an announcement?
On Sun, May 16, 2010 at 10:15 AM, Bruce Ferrell bferr...@baywinds.orgwrote:
I'm trying to make an AMI call. I want to call a number, play an
announcement when the call is answered, then call a second number and
connect the two
Hi Guys,
Running the following with a Sangoma A101D PRI card:
*Asterisk 1.4.21.2*
*LibPRI version: 1.4.10*
No inbound or outbound calls can be made. In fact Asterisk CLI doesn't show
any activity. Problem goes away on restart of the system or maybe asterisk.
I see post about upgrading Libpri to
Hi Guys,
I have to upgrade to latest Libpri 1.4.10.2 due to an existing bug in the
current 1.4.10 version. I am running Asterisk 1.4.x (in fact it is a
PBXinaFLASH system).
How can I upgrade to the latest Libpri? Do I need to re-install Asterisk?
Won't that break the box?
Can I simply do this
17, 2010 at 3:48 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, May 17, 2010 at 03:22:04PM -0400, bruce bruce wrote:
Hi Guys,
I have to upgrade to latest Libpri 1.4.10.2 due to an existing bug in the
current 1.4.10 version. I am running Asterisk 1.4.x (in fact
/lib ; ln -sf libpri.so.1.4 libpri.so)
install -m 644 libpri.a /usr/lib
if test $(id -u) = 0; then /sbin/ldconfig -n /usr/lib; fi
Thanks,
Bruce
On Mon, May 17, 2010 at 4:03 PM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the help Tzafrir.
I think for libpri you meant = 1.4.x rather than
That is the RFC number for SIP. Yes, Asterisk is compliant with RFC. I am
not sure to what degree but I haven't ever faced non-compliance on SIP RFC 3261
ever with any provider.
-Bruce
On Wed, May 19, 2010 at 2:28 PM, Tarek Sawah tareksa...@hotmail.com wrote:
Greetings List,Trying to
Is the Java soft phone an open source or obtainable? I am just checking
their site and it seems they only provide service??!!
Their java web based client is built neatly. Would like to test that on my
servers.
On Thu, May 20, 2010 at 3:21 PM, mgra...@mstvp.com wrote:
I've used HP Thin Clients
Thanks for the update. How to upgrade to the latest stable release without
compliling Asterisk again? Can you please explain and detail the commands?
We are running PBXinaFlash with LibPRI 1.4.10.1 which gives lots of
problems.
Thanks
On Wed, May 26, 2010 at 12:27 PM, Asterisk Development Team
to
strip or hide the CLID if Callee requested private presentation?
Thanks
On Sat, May 15, 2010 at 4:14 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
We have a problem with Caller ID not being displayed. I want to test
everything to see where the problem is with the incoming Caller ID
Hi Guys,
I am running a PBXinaFLASH server. I replaced contents of /usr/src/libpri
with the new version of Libpri v1.4.11. The installed one was v1.4.10.
System is running Asterisk 1.4.21.2.
I did the following after:
cd /usr/src/libpri/
make
make clean
make install
Install end with these
:
- bruce bruce bruceb...@gmail.com wrote:
What am I doing wrong that it's not update to 1.4.11?
Thanks, Bruce
--
Did you restart your services to ensure the new library was picked up?
--Tim
--
_
-- Bandwidth and Colocation
Hi Guys,
Anyone else can comment on this or give me their thoughts please? I just
want to know if someone can confirm the output for make install in new
LibPRI directory.
Thanks,
Bruce
On Fri, May 28, 2010 at 12:58 PM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the input. Yes, I did
Hi Guys,
I am looking to use System() function along with some bash scripting to
determine if a Trunk is being used during certain time of the day or not.
Here is what I have in mind. Please guide me if you know a better way:
exten = s,1,answer
exten = s,n,System(/tmp/check.sh)
check.sh:
check
be really helpful.
Thanks,
Bruce
On Sat, May 29, 2010 at 5:28 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Should be solid. After all munin also works on the same lines and it works
solid.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-05-29 5:12 PM, bruce bruce
?
Thanks
On Sat, May 29, 2010 at 7:07 PM, Steve Edwards asterisk@sedwards.comwrote:
On Sat, 29 May 2010, bruce bruce wrote:
I am looking to use System() function along with some bash scripting to
determine if a Trunk is being used during certain time of the day or
not. Here is what I have
, May 30, 2010 at 9:37 AM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the tip. I have been checking those two options. Would you be
able to provide an example of how GROUP or GROUP_COUNT may check for a
trunk
usuage?
Here is how I do it. It is based on Asterisk 1.6.1.x, and I created
know, in 1.6 is no more call-limit in sip.conf
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
bruce bruce wrote
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
bruce bruce wrote:
Thanks for the advice, but I have to keep the customer on hold till the
line becomes available
Hi Guys,
Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When
trying to dial a number, I get this:
tel*CLI Use of uninitialized value in hash element at /var/www/html/panel/
op_server.pl line 3367.
Use of uninitialized value in concatenation (.) or string at
Hi Guys,
I have tried every single rule I could into iptables but I can't register
this VPS to a provider Spikko. Finally I did an iptable accept on INPUT,
OUTPUT, and FORWARD, for ports 0:65000 just to test things and still I can't
register to the provider.
I can easily register to another
for the input.
On Sun, Jun 6, 2010 at 4:47 PM, dotnetdub dotnet...@gmail.com wrote:
On 6 June 2010 19:48, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
Just did an Asterisk 1.6.x (repo install) and FreePBX (source install).
When trying to dial a number, I get this:
tel*CLI Use of uninitialized
, Tilghman Lesher tles...@digium.com wrote:
On Sunday 06 June 2010 13:46:49 bruce bruce wrote:
I have tried every single rule I could into iptables but I can't register
this VPS to a provider Spikko. Finally I did an iptable accept on INPUT,
OUTPUT, and FORWARD, for ports 0:65000 just to test
? Are
there any errors in the asterisk logs? Does asterisk stay running after it
starts?
~Seann
On 6/6/2010 5:00 PM, bruce bruce wrote:
Reboot like 10 times and the problem still presists.
Also, upon reboot despite having done chkconfig --add asterisk asterisk
still doesn't load automatically
macro-vm, extension vmx!*
Thanks,
Bruce
On Mon, Jun 7, 2010 at 3:29 PM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, 7 Jun 2010, bruce bruce wrote:
CentOS 5.4 and asterisk does stay running after it's loaded by asterisk
-g. But the chkconfig --add asterisk doesn't work :(
What does
I did see the TTY=9 on the third or fourth line but commenting that doesn't
help much. I would really appreciate it if you can send the changes you
made.
Indeed it is a VPS.
Thanks,
Bruce
On Mon, Jun 7, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote:
*chown: cannot access
Since you mentioned FreePBX, unfortunately, it's not only the GUI that
drives the system and it can be that at some point someone planted
the extension in one of your .conf or other file if they had access to SSH
or some other way.
Going back to occurrence in sip.conf as mentioned, of course
# if test x$CONSOLE != xno ; then
# ASTARGS=${ASTARGS} -c
# fi
#fi
On Mon, Jun 7, 2010 at 8:12 PM, bruce bruce bruceb...@gmail.com wrote:
I did see the TTY=9 on the third or fourth line but commenting that doesn't
help much. I would really appreciate it if you can send
Hi Guys,
I have Spikko setup as provider of DID and outbound routes and I can make
calls out but no inbound calls via DID can be made. I did a sip debug which
is reported below. I never receive the call though, I have a catch all in my
inbound routes and it doesn't hit my context at all or not
Hi Everyone,
I have a php file that if an argument is passed to it, it will echo a number
back. I am looking to use system() in dial-plan to send ${EXTEN} to it and
then to get that processed value back from the php file and put it in $var
back into asterisk dial-plan. While trying this method
, Jun 14, 2010 at 12:12 PM, Carlos Chavez cur...@telecomabmex.comwrote:
On Mon, 2010-06-14 at 12:00 -0400, bruce bruce wrote:
Hi Everyone,
I have a php file that if an argument is passed to it, it will echo a
number back. I am looking to use system() in dial-plan to send
${EXTEN
-0400, bruce bruce wrote:
Hi Carlso,
Thanks for the input. I have done this in php and am not familiar with
phpagi.
So, there is absolutely no way to temporarily solve this problem by
getting the value back from php file?
Wondering if it would require a lot of work to change
to dialplan
exit(0);
?
Thanks,
Bruce
On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez cur...@telecomabmex.comwrote:
On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
Hi Carlso,
Thanks for the input. I have done this in php and am not familiar with
phpagi.
So, there is absolutely
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