you could try to set a var to the exten maybe.. and then use that var ..
since when in exten = i , well i will be the exten..
On Tue, Apr 15, 2008 at 11:52 AM, Anonymous [EMAIL PROTECTED] wrote:
Originally posted by: mailto:
Hi all
Now I'm making IVR sequance that is customised
How do you get 11ms translation time on ulaw 729 ?
we have 12ms and its dual xeons 2.6..
On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote:
Ok, I built a test system to duplicate my problem and provide myself
a platform that I can mess around with to try and break any features.
My problem is
We can port most of these numbers, give us a call to see how fast we can
switch this over,
Meanwhile we know Les, so we can ask them to push temporariliy to our
switches while it's being transfered.
On 8/9/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
Please stop advertising your
is subscribe context an addiotional switch/field ?
or its the peer context ?
On 8/9/07, Mike [EMAIL PROTECTED] wrote:
I feared so, but I have already started working on this. Thanks for the
confirmation.
Too bad, the rest of my design was relatively elegant (IMO) and easily to
modify.
hmm from what i have seen this is not supposed to be.. the info is still
there but should not be used in case of privacy..
zap show channels always show last info till a span refresh.. but the
privacy should indeed replace those with Privacy.
Maybe it could be a bug ,
On 8/9/07, Jeremy Mann
yes we have several of these...
you need to use a PCI expensiont slot.. its an addon that piggy backs to a
blade and takes 1 u ..so total blade will take 2 u's...
but you can hook 2 PCIS on it.. sangoma or whatever...
This way we can Redundantly failover 2 PRIS on each other with each blade
http://outcall.sourceforge.net/
we use outcall
and modded the source directly for our apps.. 0$ fee.. 100% flexibility..
Works like a charm !
On 6/6/07, Martin Smith [EMAIL PROTECTED] wrote:
We've been using SIPTAPI and love it for our call center. We originally
used ASTTAPI, but liked the
yes on home pbx i love the s/CALLERID..
maybe you should
f($[${CALLERID(number)} = 15552221313]?15:5)
try to isolate string to strings.
this is not good i think
you need qhotes on the callerid part too if you evaluate to the 1555xxx
f($[${CALLERID(number)} = 15552221313]?15:5)
maybe im
wget -q -O - --connect-timeout=5 http://www.voip-info.org |grep '149461'
gives me the string..
its up for now.. could of been just rebooted.
On 6/6/07, Roger Schreiter [EMAIL PROTECTED] wrote:
Ed Nuñez schrieb:
Is anyone else having trouble going into voip-info.org today?
Yes. Me.
also note vnaks are iax i think
On 6/6/07, Henry Cobb [EMAIL PROTECTED] wrote:
On 6/6/07, Matt [EMAIL PROTECTED] wrote:
I chart VNAKs per hour.
Would you care to share how you accomplish this? What programs do you
use?
grep VNAK /var/log/asterisk/full | cut -d ' ' -f 4 | cut -d : -f 1 |
what versions of asterisk on both systems ?
On 6/5/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
Have you tried something along the lines of:
System(swift blah blah blah -o blah.wav)
Playback(blah.wav)
It does have an inherent delay for the generation step but maybe swift
means (I)f (I) (R)emember (C)orrectly
On 6/4/07, ram [EMAIL PROTECTED] wrote:
This notifies you that it has been used (IIRC).
Hi
what does that mean , it has been IIRC ?
ram
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that becasue the reinvite is using a private ip probably..
sip debug
pastebin the results..
look in the re-invite part..
On 6/4/07, Compnet Bobby [EMAIL PROTECTED] wrote:
We have the latest version of asterisk, on a xeon dell server (2gb ram),
with 6 snom320's(latest firmware) and 3
i would force a timer on it..
dial (blah,30)
maybe that would bypass , maybe not..
i actually think it wont..
another example of this problem is DNS
echo '1.2.3.4your.favorite.itsp' /etc/hosts
then Dial(SIP/[EMAIL PROTECTED])
DNS failing will BLOCK the call indefinitely...
On
may i add , eyebeams confnig file is xml and could be generated , BUT, the
password is hashed in some way.. any idea on that ? its a pretty long hash
On 4/25/07, Senad Jordanovic [EMAIL PROTECTED] wrote:
Andrew Furey wrote:
On 24/04/07, Senad Jordanovic [EMAIL PROTECTED] wrote:
Tzafrir Cohen
echo cancel all the way, any company not including that in first place is
just selling a car without the wheels..
i would see the a card without echo cancel as driving in winter with summer
tires..
On 4/23/07, Erik Anderson [EMAIL PROTECTED] wrote:
On 4/23/07, Tom [EMAIL PROTECTED] wrote:
From overall apprecation feedback :
#1 Polycom (Any)
#2 Aastra 480i
#3 Cisco 7940+
#4 Linksys SPA-94x
On 4/11/07, Stephen Bosch [EMAIL PROTECTED] wrote:
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
I'm
No entirely true.. but yeah realizing we are talking about asterisk .. you
can't do reliable T38 faxing..
check out openpbx its embeded , all you need then is a ITSP to do TDM
termination as T38
A bit more expensive then voip.. but you still get the bulk /automation part
without the
No you are being misled.. SER can NOT DO IAX, SER = SIP only
but you would need SER to do that yes.
On 4/12/07, Alex Balashov [EMAIL PROTECTED] wrote:
Certainly. Any signaling / trunking protocol will do, in principle.
On Thu, 12 Apr 2007, Edgar Guadamuz said something to this effect:
tried x+102 ?
On 4/5/07, Brent [EMAIL PROTECTED] wrote:
I'm trying to get an IAX trunk to failover to a local trunk it the trunk
is down.
This is what I've been working on:
[macro-forward1];
exten = s,1,Dial(IAX2/192.168.1.1/${ARG1},20)
exten = s,2,Goto(call-${DIALSTATUS},1)
exten =
sip would be the required one as iax..well..
also openwengo wont work.. to much overhead .. broswrer needed.. ie
component + flash + css+js etc.. not viable..
so im also asking anyone have one ? since ihave a supply of around 2000 of
the vonage usb stick OEM..
On 3/30/07, Michael Van Donselaar
From what i was told via the GT guys at bell
The customer should send
I 1C Standard Facility Length = 21
9F Serv Discrim Networking Extensions
PDU Component Begins (hex)
8B0100A10F020101...
instead of:
I 28 Display Length = 9
E Display .JIMBOB
so we
the rest to talk on a specific topic and help each other out...
On 3/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Mar 03, 2007 at 11:52:38AM -0500, Mike Lynchfield wrote:
From what i was told via the GT guys at bell
The customer should send
I 1C Standard Facility Length
Please note that we are available to fix the current REMOTE crash that
affects Asterisk/openpbx/trixbox and crashes these systems via a malformed
packet
please contacts use if you need a hand to patch your systems.
--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP
);
return -3;
}
is my patch
We just offering to support people that don't know how to mod stuff and
recompile..
On 3/2/07, BJ Weschke [EMAIL PROTECTED] wrote:
On 3/2/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
Please note that we are available to fix the current REMOTE crash
nice one.. we have rogers and primus.. ni'2 and same..
let me know if this ni2 and ni1 thing is crap or not
On 2/28/07, Webster, Andrew [EMAIL PROTECTED] wrote:
Outbound calls on my Telus PRI aren't taking the Name portion of the
callerID. I've looked at the logs, and it is being set (see
We have decided to allow our tech's to do support for non-clients of
voicemeup.com
You can head to http://support.voicemeup.com/ and one will be in touch 8 to
6pm business hours.
3 levels of support are offered for Asterisk/compiling Trixbox , Ivr's etc.
--
Mike
Sales Manager
try not using dst.. maybe its a regex on te fieldname that matches for
reserved keywords..
try pre_dest instead
On 2/28/07, Bayrouni [EMAIL PROTECTED] wrote:
Hello,
I created a new field named pre_dst of type varchar(80) exactly like dst
field in cdr table.
In the dialplan I put:
exten =
try putting near the exten = 1000,1,dial stuff
On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
I've installed Sven Slezak's Notify module. He gives the follow as an
example line to put into extensions.conf
exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/
sunnybook)
I
You could try Fast agi.. then i think master agi deamon runs from services
and replies to requests by including sub scripts.
however i do see some connect failures sometimes...
On 2/28/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Feb 28, 2007 at 10:56:14AM -0800, Yuan LIU wrote:
you need also note that once you do that asterisk authorizes on first ip it
sees in sip peers..
so client a and client b with same ip.. could cause problems unless you
divide them .
On 2/28/07, Bayrouni [EMAIL PROTECTED] wrote:
Eric ManxPower Wieling a écrit :
Yuan LIU wrote:
From: Eric
or at http://asterisk.voicemeup.com for consolidated lists..
On 2/28/07, Rodrigo Gonzalez [EMAIL PROTECTED] wrote:
Bayrouni wrote:
Sorry for disturbing, but I sent some messages today and I am not seeing
them on this list.
Can sombody tell me, in case this message appear on the list.
Just to let you all know we are offering 0.019 down to 0.008 automatic
pricings on volume..
TDM termination/origination
Unlimited SIP/IAX accounts
g729/ulaw/alaw/gsm/etc
15 channels opened per account to start with
Toll Free numbers / Local numbers
Reseller rates
Wholesale Rates
Whitelabel
Do You have a link to ACT CRM ?
Thanks
On 2/20/07, Cory Andrews [EMAIL PROTECTED] wrote:
Has anyone ever been party to an integration of ACT CRM platform with
Asterisk?
Thanks
Cory Andrews
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Well, could be the fact provider not pushing as g729 or someting else.
Can you set debug 999 and set verbose 999
then redump that ? you are missing the before the answer part also..
Also try G711 first then work your way to other codecs
On 2/20/07, Rajeev Natarajan [EMAIL PROTECTED] wrote:
Yes first thing is not using 1.4 but as you probably won't budge , try
hints.
exten = 1001,hint,SIP/USER
that will force it to poll status of that peer and reset the queue agent, of
course replace values with actual ones
On 2/20/07, Paul Hales [EMAIL PROTECTED] wrote:
Are you using
With all other things said.. you might want a professional service for this
like targusinfo.com
Maintaining and running an operation like a cname web lookup thing is REALLY
high overhead in terms of web traffic etc
What happens when you get 30 ITSP/clients pulling 1000 calls each or 10
Well caching is the way to go., bu then again most of the current solutions
have this problem.
John smit has a DID.. 514 555 1234 and closes account.. did sleeps for 3
months and new client Jane doe takes it..
Now how long should caching be ? this is a big problem ATM because some
cache for 1
You can try us, http://www.voicemeup.com
TDM in most areas , others offloaded white routes to L3 mainly.
Cover most of usa , and canada.
you can ping www.voicemeup.com to get an idea on location , we are directly
on peer1,teleglobe,videotron with best quality bandwith only.
Per minute pricing
also trixbox stop registering randomly on all versions..
confirmed with over 200 client accounts over here...
all using trxibox.. asterisk vanilla is ok
On 1/31/07, Alejandro Lengua [EMAIL PROTECTED] wrote:
How many simultaneous calls per account are you sending ?
On 1/31/07, Peter
Just to let you know all we consolidated all posts on
Asterisk/openpbx/freeswitch into 1 forum for ease of viewing.. threaded of
course..
http://asterisk.voicemeup.com
--
Mike
Sales
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
___
reboots are wiseOn 10/16/06, Tom Vile [EMAIL PROTECTED] wrote:
fine for me here since it came out. We are running 15 extension all day long.On 10/16/06, shadowym
[EMAIL PROTECTED]
wrote:I am getting ready to image a production system.Right now I am planning on
using Centos 4.4, Asterisk
sounds like itts missing the mysql -source libsOn 10/13/06, raviprakash sunkara [EMAIL PROTECTED]
wrote:Hello Users,I Installed the Asterisk-1.2.11, For My Real time Use I'm Use MySql For Asterisk Database, By Using the Asterisk-addons -
1.2.4 in My Linux.For My Voice messages Storage , I want To
Look into DeadAGI, should be easy enough that illl implement tomorow ;)On 10/13/06, Klaverstyn, David C
[EMAIL PROTECTED] wrote:
Can this be done?
I call Asterisk using my mobile (cell), Asterisk then hangs
up on me so I am not charged for the call. Asterisk then calls my mobile
Anyone ?On 10/12/06, Mike Lynchfield [EMAIL PROTECTED] wrote:
Can anyone share sysctl tuning params for asterisk and unix ?trying to see if we have differences in them -- MikeSales Manager
http://www.theclubvoip.com
Making it happen1.877.807.VOIP (8647)
-- MikeSales Managerhttp
Want to share these 13 packets ?On 10/13/06, Matt Loretitsch [EMAIL PROTECTED] wrote:
http://www.elna-america.com/tech_al_reliability.php
Capacitors are one of the components on that motherboard that have afinite life span.Other components are more or less tolerant of thesechanges over
Can anyone share sysctl tuning params for asterisk and unix ?trying to see if we have differences in them -- MikeSales Managerhttp://www.theclubvoip.com
Making it happen1.877.807.VOIP (8647)
___
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yes.. actualy use 1 did for each proxy to check..then inbound for each use the method he described..On 10/12/06, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
on an analog Zap PSTN channel, you have no real way of determining ifthe remote side answered, because, as you discerned, it IS
Actualy to have fun.. make it a playback(dialtone),300 ;)that will make your little hacker think they have a dialtone then Record the number dialed and put that in a db for further investigation..Actualy could also be a user that has no clue on how to configure the system.
On 10/12/06, Eric
yeah...I Got a Siemens Phone and i can't hear the ringing.
Try to change these settings in the pap2 device (Admin - Advanced Mode-Regional settings)
Voltage = 90V
frequency = 20 Hz
impedance = 900 ohms
waveform = trapezoidal not sure about the question but this is a must i
hehe yeah.. still when you see that qualify breaks newer xlites' you would wonder why to use it anyhow ?On 7/11/06, Rick Smith
[EMAIL PROTECTED] wrote:teliax had a 2.5 hour outage today. I wouldn't call that short.
-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]
] On
#1.. most the failures and network bottle necks on asterisk in a 1k + user sip /iax are registrations polling'syou are right .. get SER ... dont be dumb.#2 the config file with asterisk hardcode ips is a simple matter of running a script that parses it and puts in whatever it needs
#3 basic
something that you could drill into.. or even search.. hold on mate i got this..how about a master LCR system that would generate config for users in terms of filters..EX1: filter on qoswould return BEST QOS list of all terminations for providers
like..provider1=sip/[EMAIL
larry each of these files do something for a specific needs.. hence the sip.conf is for sip related modules..iax etc etc..voicemail.conf if you need voicemailres_odbc etc for any database usage..
basically read the manual and look into each files to see what they do..asterisk will start and work
trxtel ping me.On 7/11/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Tue, 2006-07-11 at 14:10 -0400, Rick Smith wrote: teliax had a 2.5 hour outage today. I wouldn't call that short.its all relative, nufone had a 30 day outage :P--Trixter
http://www.0xdecafbad.com Bret McDanelBelfast
Message- From: Aaron Daniel [mailto:
[EMAIL PROTECTED]] Sent: Tuesday, July 11, 2006 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Server redundancy On Tue, 2006-07-11 at 14:34 -0400, Mike Lynchfield wrote:
#3b.. we need multiple listening addies
if we use call out files in asterisk it only creates a cdr on the bottom leg or the callfile.ex: will have a cdr entry for the channel : but not the extension ;Anyidea how to fix that behaviour ?
-- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.888.470.7253
ok, We are building the perfect voip company..we are trying...we need input on end-users:reply to my email with --ENDUSER in subject.with anything you would like to see your current voip provider offer online/offline ( don't say.. support, an answer on phone etc) be constructive..
reply to myt
can you elaborate on modify sip to update the status on the sip friends in realtimethanksOn 6/29/06, Doug G
[EMAIL PROTECTED] wrote:What I did was modify sip to update the status on the sip friends in realtime. Then via FAGI dial them directly with the data found in real-time. (ie dial (
HERE IS answer [EMAIL PROTECTED]had same problem..make the settings for 90 volt.. not 70 volt ringer.. make it trapezoidal not sinusoisalmake it 900 ohm not 600 impedence..that worked for pap2's
seem siemens are made for europe style ring voltage not north american.On 6/27/06, Herchi Silviu
We use cisco 7960's but thats not cheap..BTW Doungyour signature :Those that sacrifice essential liberty to obtain a little temporary safetydeserve neither liberty nor safety. -- Ben Franklin (1759)
is a good one.. tell that to your president..and the patriot act.s/patriot/cutallrights/PS Andrew..
BLAH=1BLAH=1On 6/27/06, Brian Capouch [EMAIL PROTECTED] wrote:
Jon Schøpzinsky wrote: Hello As far as ive understood, you can just write
Exten = s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and ${AVAILSTATUS} would return 1Through more testing, the double quotes
RegardsJosué
2006/6/27, Mike Lynchfield [EMAIL PROTECTED]:
HERE IS answer [EMAIL PROTECTED]had same problem..make the settings for 90 volt.. not 70 volt ringer.. make it trapezoidal not sinusoisalmake it 900 ohm not 600 impedence..that worked for pap2's
seem siemens are made for europe style ring
dual support yes.. however i read a few articles on the fuct that single with double the ram is better..something about the bus or sshare between both processors i think.i would go AMD opteron, but that me.
or sunOn 6/13/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
HelloIs it correct that IAX2
taskset does not seem to exist on redhad 9 nor freebsd..;)On 6/13/06, Zoa [EMAIL PROTECTED] wrote:
When i did this test ages ago, i found out that iax was worse than sip,but sip was worse than trunked iax.
Joachimolin Anderson wrote: I use IAX2 quite a bit and I haven't really noticed any
. They want their operating system back. :- )
-Original Message-From: Mike Lynchfield
[mailto:[EMAIL PROTECTED]]Sent: Tuesday, June 13, 2006 9:42
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] IAX2 Vs SIP cpu
loadtaskset does
how baout codecs ?try enabling all for testing ..then limit..On 6/9/06, Jason Lixfeld
[EMAIL PROTECTED] wrote:Kinda confused by this...I have a Cisco 7960 configured with a
couple SIP extensions configured on the phone.Just trying to dialone extension from the other on the same phone, but when I
could it be IPP VS digium implementation ?On 6/8/06, William Piper [EMAIL PROTECTED] wrote:
Send an email to
support@plainvoip.com. They are normally quite helpful.
bp
On 6/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote:
Do you have the g729 codec? On 6/8/06, Henry J. Cobb
[EMAIL PROTECTED]
yes you need to waiti assume you are using latest ?WAIT(5) should work..also i guess you need an answerOn 6/9/06, Lacy Moore - Aspendora
[EMAIL PROTECTED] wrote:
Mine has usecallerid=yes and caller id works. Not sure if that's the problem or not.
On 6/9/06, Curt Shaffer [EMAIL PROTECTED]
wrote:
well.. whats the question...if you shut down an essential servic eof course you will have this..its like saying i removed the drive and all crashed.On 6/9/06,
Andrew Kirch [EMAIL PROTECTED] wrote:
At approximately 3:15pm I shut down the office MySQL server to changeout some hardware.Shortly after
but what if the ast_log function is conditioned ast_log(LOG_WARNING, Autodestruct on call '%s' with owner in place\n, p-callid);whats in there ?On 6/9/06,
BJ Weschke [EMAIL PROTECTED] wrote:
On 6/9/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jun 08, 2006 at 10:44:39PM -0400, BJ Weschke
if on freebsd..stop asteriskkillall mpg123cd /usr/ports/audio/madplay/make make installedit musiconhold.conf[default]mode=customdirectory=/usr/local/share/asterisk/mohmp3
application=/usr/local/bin/madplay -Q -o raw:- --mono -R 8000 -a -12then restart asterisk.. Mikehttp://www.theclubvoip.com
On
cisco topic .. is there a sip image for 7980's yet ?On 6/7/06, Aaron Daniel [EMAIL PROTECTED] wrote:
You have to press settings, then **#, and wait a moment to make sure itunlocks.Then you can configure a tftp server to use.
The alternative is to configure your dhcp server with a tftp
in Fact we saw similar problems with all sipura products. We think its a default value thats not quite right for the north american market, these units are built and tested in asia mostly.one simple test to check it out is call this number
www.nextwavetitaniumplus.com Toll-Free Account
i got cepstral loaded and cmu recognition server waiting for comands..i instaled basic rezrov and got my main screenhow do we enable the speech onto it ?On 6/6/06,
John Todd [EMAIL PROTECTED] wrote:
http://www.boingboing.net/2006/06/05/play_zork_by_phone.htmlLet me preface this idea with one
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