make sure that the shell location is right...#!/bin/bashnot equal to #!//ust/local/bincommon mistake.. On 5/25/06, Maxim Vexler
<[EMAIL PROTECTED]> wrote:On 5/24/06, Michael Collins <
[EMAIL PROTECTED]> wrote:> > you should mv the file (and in the same filesystem, so 'rename' is> used)> >>> You mi
in a dialing plan.. extensions.confcan we enable or force a codec on specified npa..EX: 514NNN,1,force(gsm) 514NNN.2. dial(sip/blah)???
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
does one know how to program so i can have 2 lines on one sip account on that phone ?im runnign my own asteriskdo i need 2 local accounts ? one for each line ? that rebounds to same SIP forp VOIP provider ?
___
--Bandwidth and Colocation provided by Easy
Ok anyone have latest cdrtool running 4.1 i think..
ill pay for install
On 1/31/06, Jimmy Smith <[EMAIL PROTECTED]> wrote:
i understand.. anyone know how much is basic support from them ?
On 1/31/06, [EMAIL PROTECTED] <
[EMAIL PROTECTED]> wrote:
Hello,Call
sip:[EMAIL PROTECTED]
i understand.. anyone know how much is basic support from them ?
On 1/31/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Hello,Call sip:[EMAIL PROTECTED]
Regardsharry--- Jimmy Smith <[EMAIL PROTECTED]> a écrit :> anyone having weird problems on latest cdrtool?>>> #!/usr
anyone having weird problems on latest cdrtool?
#!/usr/bin/php4
Fatal error: Class webservice_ngnprocdrtool_ngnprocdrtool: Cannot inherit from undefined class soap_client in /var/www/CDRTool/SOAP/client_lib.php on line 2
always get weird error like that
___
SIP/localext
or ZAP/someothr
or providers SIP/1xxxnnn
On 1/18/06, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
With most asterisk installs, there's no difference between an extensionand a phone number. For example, from your internal context, a phonecould dial one number and get a
i only see the spa1001 on voip-supply
http://www.voipsupply.com/product_info.php?manufacturers_id=14&products_id=320
can you confirm this is what i want ?
how would it work ? the phone jack on spa to the house patch and the rj45 into the asterisk hub
?
On 12/23/05, Ben Higley <[EMAIL PROTECT
but i could connect it to the home patch box and use any phoen in house right ?
On 12/23/05, Ben Higley <[EMAIL PROTECTED]> wrote:
the FXO is to connect to PSTN..You need an FXS device (like a sipura 1000, 2000, or even 3000), or aniaxy. etc../Ben> I got asterisk at home and not ma Bell, so i inte
I got asterisk at home and not ma Bell, so i intend to use the internal
house wiring to use and connect a patch cable to my asterisk to the
house.
this way i can pick up any extension in hous and will pop on zap device..
now
i got a x101p
Dec 19 09:26:07 kernel: wcfxo: DAA mode is 'FCC'
Dec 19
ahaahah now thats something to be worry about.. that prolly coz they
dont want to pay taxes and your invoices serve you as refunds/credits
for irs..
BUT it is required by law to give irs that crap..
so i guess they pushing in offshore an will disappear someday with all the service and cash.
Whe
FYI 60/1 measn first 60 seoncd billed then each 1 /60th of a minute
so 1minute 25 second call is billed as 1 minute 25
45 second as one minute
wher the first number is minimum seconds
so 6/6 is first 6 seconds no matter what then every block of 6 seconds..
6/1 well you get the point..
its lik
maybe your account was disbaled due to non payment ?
also check if you are sending auth in the features area..
On 12/22/05, Jonathan k. Creasy <[EMAIL PROTECTED]> wrote:
This is an authentication problem. Check the username, password, numberand context being sent across to see if they are correct
hey 1.2 b2 hs bugs.
On 10/18/05, Adam Moffett <[EMAIL PROTECTED]> wrote:
- Original Message -> *From:* Rob Fugina [EMAIL PROTECTED]>> *To:* asterisk-users@lists.digium.com
> asterisk-users@lists.digium.com>> *Sent:* Monday, October 17, 2005 5:14 PM> *Subject:* [Aster
yes i got my mainstream * with teliax
no problem..
keep it ,ulaw,alaw, no jitter and ,open your ports, all good for 9 months !
On 10/19/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
For those on the list using iax with teliax.com, the intermitant one-wayaudio problem that I reported to them re
i use teliax for primary and bell for failover.
no neeed yet for failover ;)
they got theyre shit together...On 11/3/05, Jason Brashear <[EMAIL PROTECTED]> wrote:
Thank you Gleim I will look into that.
-Jason
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf
oh and one tech from them support said they will be handling all cname soon .. like ETA 1 week
can't wait to get my FBI name out...
On 11/5/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
> OK I am exhausted.> I can't seem to figure out how to send a caller ID along with a> Outbound call. < snip >>>
bet its a difference in register timer..
On 11/5/05, John Fraser <[EMAIL PROTECTED]> wrote:
Hi all, I am using a TE405P with an E1 from the telco. I am getting the allcircuits busy now message about 5% of the time on outbound calls from sipsoft phones. Has there ever been a resolution to this pr
my bad you are.. lol didnt realize..
On 11/4/05, Jimmy Smith <[EMAIL PROTECTED]> wrote:
you could wait infinitely or try users list..On 11/4/05, harry gaillac <
[EMAIL PROTECTED]> wrote:
Hello Walter,The ser an asterisk run in the same box.What do you mean "redirect host + po
you could wait infinitely or try users list..On 11/4/05, harry gaillac <[EMAIL PROTECTED]> wrote:
Hello Walter,The ser an asterisk run in the same box.What do you mean "redirect host + port :)"
Sip agents send sip requests to ser (outbound proxy)and this one to asterisk !sip agents are both registe
seems every 10 sec something is happeneing on your network...
make sure your router is rebooted often if you have QOS on it has they tend to get behind on queues..
or UDP crc checksum failing in router.. that happened to me
on a linksys
your ping is ok 60 is good
i would also test my lan qualit
running 1.2 on both servers
from A to B to a 7960
the 7960 receives callerid as
"NAME
usernameofsipuser"
i tried setting callerid etc before doing the dial
A to B is via iax B to 7960 is via sip of course
on B right before i dial 7960 i noop calleridnum and name and both populated ok
is th
hey all.. got a nice one.. got a cisco phone connected to
asterisk A .. withc connects to ASTERISK B ... my VM
is on B.. is there a way to relay VM notif to cisco ?
thanks
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
hehe yes exactly..
you could tail -f that file .. or grep
as in
tail -f /var/log/asterisk/verbose |grep -10 -v 'somestring'
that would give you 10 lines around it.. or before it i dont remmeber
off the bat..
On 8/24/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote:
> Matthew Boehm wro
line1_shortname: "Home"
line1_displayname:"Home"
On 8/24/05, Asterisk User Group <[EMAIL PROTECTED]> wrote:
> I have three questions about my 7960 phone that I can't discern from the
> docs/wiki.
>
> 1st - If I change the SIPxx.cnf file to change registrations it sets
> up new lines as expec
or verbose..
makse sure its enabled in logger.conf
On 8/24/05, Jimmy Smith <[EMAIL PROTECTED]> wrote:
> how about /var/log/asterisk/mssages
>
> On 8/24/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> > We have recently started routing about 3 PRI's worth of t
how about /var/log/asterisk/mssages
On 8/24/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> We have recently started routing about 3 PRI's worth of traffic thru our
> asterisk box.
>
> The text on the console now flys by so damn fast, I can't really see
> what the heck is going on. Even with verbo
mae sure you dont have skip-networking in my.cf for mysql
also make sure you can connect to ip , name and localhost form cmd
line first using these credentias
On 8/24/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> John Novack wrote:
>
> >> Both modules should be able to use either hostname
Anyone got
Realtime with mysql database using myodbc-iodbc
and getting errors like SqL FETCH ERROR on load ( 50 q /s ) on db ?
my db has 500 threads ready
this happens sometimes.. not always
im not sure if its asterisk -> odbc connector
or
odbc connector to mysql
but i assume its the first
Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats
is from what you pasted btw..
Don't know any of 0x8 formats
is
524288 (1 << 19) (0x8) videoh263 (H.263 Video)
meaning it downst understand it or find it
On 8/17/05, Jimmy Smith <[EMAI
quickly this looks like a incompatible codec.. or unrecognized..
show codecs on CLI>
show show
262144 (1 << 18) (0x4) videoh261 (H.261 Video)
524288 (1 << 19) (0x8) videoh263 (H.263 Video)
1048576 (1 << 20) (0x10) video h263p (H.263+ Video)
does it ?
O
pruning breaks asterisk on high loads
at least on all 5 of our servers.
all using different versions and custom.
"
What you can do is use "sip prune realtime " to remove just the
single peer/user from memory. And you can force a reload of that peer
from realtime by using "sip show peer lo
coudnt agree more.. thats exact thing i was saying the other day..
please hold my di..k while i take a leak i don't want to wet my hands.
RTFM, google and test. || Pay
On 7/6/05, Brian West <[EMAIL PROTECTED]> wrote:
> Why not do your research instead of asking the list to do it for
> you l
i can suggest using wavepad.
its on the voipinfo site
On 7/6/05, mohammad <[EMAIL PROTECTED]> wrote:
>
> HI ALL;
>
>
> I have problem converting a windows .wav file to .gsm format by Sox.
> Could anyone help.
>
>
> Cheers,
> Mohammad
>
>
does cdrtool handle 800 termination from different src ?
from the page they say
"Combined rating based on traffic, duration, application type and destination"
so not from src it seems..
anyone got that working ?
Example : billing depending on src number + destination #
JV
___
does cdrtool handle 800 termination from different src ?
from the page they say
"Combined rating based on traffic, duration, application type and destination"
so not from src it seems..
anyone got that working ?
Example : billing depending on src number + destination #
JV
___
this happens to me too.
On 7/6/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Wed, Jul 06, 2005 at 11:48:27AM -0500, Brian West wrote:
> >
> > /b
> > ---
> > Anakin: "You're either with me, or you're my enemy."
> > Obi-Wan: "Only a Sith could be an absolutist."
> >
> > On Jul 5, 2005, at 12:02
6 beyond-the-network.LosAngeles.savvis.net (208.173.57.30) 33.966 ms
34.143 ms 33.841 ms
7 * * *
hangs there...
savvis invoice paid ?
beyond-the-network a black hole ?
On 7/4/05, Gary Reuter <[EMAIL PROTECTED]> wrote:
> Hmmm
> Can't place calls...
> Can't access website...
> Neither
or use an iax client..
many are out there
http://voip-info.org/tiki-index.php?page=VOIP+Phones
On 7/4/05, Joseph <[EMAIL PROTECTED]> wrote:
> On Mon, 2005-07-04 at 17:48 -0400, Jimmy Smith wrote:
> > another example of what i was saying..
> >
> > connect directly
how about /etc/rc.local
#a line that would work
path/to/screen -d -m path/to/asterisk -vvgfc
-d -m Start screen in "detached" mode. This creates a new session but
doesn't attach to it. This is useful for system startup
scripts.
On 7/4/05, Carl
another example of what i was saying..
connect directly if still does this its not you my friend..
people should really stop using asterisk as first connect attempts to
test a service.
use a direct client on provider .,
Asterisk is complicated with many settings, unix flavors , hardware
and ba
you guys are so friggin funny..
all i see bout problems on most providers here are users who never
read a line of the handbook
i could prolly solve all these eyes closed with the asterisk handbook
on my side as a friend.
wake up..
i work for a hosting provider and we get lots of users assuming
42 matches
Mail list logo
