12 apr 2006 kl. 08.46 skrev Michael Strelnikov:
What caching DNS do you recommend?
Anyone you feel comfortable running.
/O
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visi
I have one of the Draytek USB adaptors.
Can someone point me in the right direction on how to get mISDN running with
it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, 17 March 2006 12:17 PM
To: Asterisk Users Mailing Li
Please use the asterisk-biz mailing list for all commercial
offerings. Thank you.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
___
--Bandwidth and Colocation provided by Easynews.com --
Asteris
What caching DNS do you recommend?On 4/12/06, Olle E Johansson <[EMAIL PROTECTED]> wrote:
11 apr 2006 kl. 14.28 skrev Michael Strelnikov:> I do have that line. I also have all my phones defined by IP> address. But all providers are defined by names.>> On 4/10/06, Michiel van Baak <
[EMAIL PROTECTE
11 apr 2006 kl. 16.05 skrev Brent Torrenga:
Out internet connection was out this morning. It seems that the SIP
extensions on our LAN were affected. Behavior like:
Call comes in over POTS to a TDM400P, there is a delay then before
the Cisco
79[46]0's start to ring.
If we were lucky enough t
11 apr 2006 kl. 14.28 skrev Michael Strelnikov:
I do have that line. I also have all my phones defined by IP
address. But all providers are defined by names.
On 4/10/06, Michiel van Baak < [EMAIL PROTECTED]> wrote:On
22:14, Mon 10 Apr 06, Michael Strelnikov wrote:
> Hi,
>
>My * refuse
[EMAIL PROTECTED] wrote:
I've been working on a project for Asterisk for some time and it is
finally ready for a beta release. Any feedback is well appreciated. At
the basic core it's a Dialer for Windows. I'll be adding more features
quickly, but I wanted to keep everything simple and stable in
Hi Folks,
I am running a SPA-3000 behind a legacy PABX on an analog line.
I have been able to set up a dial plan that sends outgoing calls out
to the appropriate VSP depending on prefix, and that part and the
incoming call handling works fine.
I am now trying to implement call pickup (dial 6*)
I upgraded to SIP 1.6.5, and was very happy to see that when you change the
config via the webpage or when DHCP changes it just reloads instead of doing
a full reboot. Does anyone know how to make it reload the config from the
ftp server? The menus only offer a full restart, but I'm guessing there'
I changed from a TE410P to a TE411P and fax carriers weren't detected
anymore !
I have tried everything (recompile zaptel+asterisk+spandsp ;
echocancel=yes/no ; rxgain=0/txgain=0 ; increase Wait() ; ...), nothing
worked.
The only solution that worked for me was to install and use NVFaxDetect.
HTH.
Andy Tan a écrit :
Hi Alex,
thanks for the suggestion.
Did some checks, and thought that I could set a global variable to track
the utilized bandwidth.
Wish that there are plans for support to include variables like
SIP_CODEC in other protocols.
Actually this sounds like a really nice idea
Boris Bakchiev wrote:
Are there any advantages/disadvantages to using tmpfs as opposed to
the
following method:
Matt,
Its simple. To quote the docs, "tmpfs lives entirely in the kernel's
caches"
It will shrink and grow to accommodate the files that currently on the
filesystem.
I have a private network like this:
+---+
| firewall |
+---+
|
AFAIK, it doesn't make much of a difference if all you are going to
be mainly using is the TE card. From what I've heard and seen, a
single P4 3GHz machine will handle a fully loaded TE4XX board with no
problem.
- Waldo
On Apr 11, 2006, at 10:30 PM, Tim Connolly wrote:
I was offered an
I followed the upgrade procedures from 4.1 to 5.0 using the sources.
Regards.
Cristian.
From: Tim Connolly <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussion
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] res_con
Hi Alex,
thanks for the suggestion.
Did some checks, and thought that I could set a global variable to track
the utilized bandwidth.
Wish that there are plans for support to include variables like
SIP_CODEC in other protocols.
Regards
On Tue, 11 Apr 2006 12:50:56 -0400, "Alexander Lopez"
<[EM
Dear Asterisk Users,
Just a quick reminder that the Early Bird Discount for AstriCon Europe ends on
Saturday, April 15 2006 (US Central Daylight Time). Register today for AstriCon
Berlin, AstriCon Paris or AstriCon London and save 20% ($70.00 USD) off the
standard price.
Register Now: http://www
This is very interesting, Dov has posted this question with the
subject critical, het got 7 responses, and a few of them asking him
to provide more info. But he hasn't showed again yet. I think he goes
onto /ignore
On 4/11/06, Wai Wu <[EMAIL PROTECTED]> wrote:
> You got to be kidding about 53 cal
Did you upgrade all the mysql packages, or just the server? I would bet
you missed the -dev or -lib package.
kritikus Araklidas wrote:
Hi everyone:
I installed the lates version of Asterisk with Asterisk Add-Ons. A
month ago i upgraded my database form mysql 4.1 to mysql 5.0. So after
to st
I was offered an upgrade path for my two Dell 1750's (2.8 Dual Xeon)
to get into a pair of new Dual Core Dual Opteron servers. Assuming I can
get the IRQ BS worked out so my TE411XP doesn't flip out, this should be
a pretty significant upgrade. Has anyone been able to quantify any
benefits t
I'm in Huntsville... close enough to Houston.
Aaron
On Tue, 11 Apr 2006, Lacy Moore - Aspendora wrote:
I'm in Houston.
On 4/11/06, Ryan Burke <[EMAIL PROTECTED]> wrote:
I'm interested but I'm in the Dallas area. Are there any in the Dallas
area anyone knows of?
Ryan
- Original Messa
You got to be kidding about 53 calls being recorded at sametime is an issue. I
have done at least twice as many on my dual xeon 3.4Ghz system and had no
problem as clients like to record every call that goes through the system. Then
again, in my system, the in and out channels are mixed first be
I've had a few, even on 7.4+, that were impossible to recover the
password. I usually end up looking at the current network settings and
putting an IP alias on my tftp server so it will answer the tftp get
requests coming from the phone. It gets tricky when the original config
has the TFTP s
I am East Texas outside of Marshall, if you get
something going let me know
Tom Chandler
- Original Message -
From:
Ryan
Burke
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, April 11, 2006 7:13
PM
Subject: Re: [Asterisk-Users
On Tue, Apr 11, 2006 at 02:08:09PM -0700, Daniel Hazelbaker wrote:
> Have you quit and relaunched Asterisk? (not a reload, but a full quit
> process and restart) I know in the past when I have a process
> already listening to 0.0.0.0 it will not always pick up a newly added
> NIC alias addre
The only group in Texas I am aware of is Austin. On 4/11/06, Ryan Burke <[EMAIL PROTECTED]> wrote:
I'm interested but I'm in the Dallas area. Are
there any in the Dallas area anyone knows of?
Ryan
- Original Message -
From:
Bruce Reeves
To:
Asterisk Users Mailing Li
Anybody know the proceedure to factory reset the a 7960 phone running 6.3
SIP software? I've tried holding # when booting the phone and nothing, i
can do that on my 8.2 phone but this phone i just got with 6.3 isnt working.
Also **# doesnt work either..
--
~Shaun
I'm in Houston.
On 4/11/06, Ryan Burke <[EMAIL PROTECTED]> wrote:
I'm interested but I'm in the Dallas area. Are there any in the Dallas area anyone knows of?
Ryan
- Original Message -
From: Bruce Reeves
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, A
> Are there any advantages/disadvantages to using tmpfs as opposed to
the
> following method:
Matt,
Its simple. To quote the docs, "tmpfs lives entirely in the kernel's
caches"
It will shrink and grow to accommodate the files that currently on the
filesystem.
So if you allocate 10GB for your /tm
I'm interested but I'm in the Dallas area. Are
there any in the Dallas area anyone knows of?
Ryan
- Original Message -
From:
Bruce Reeves
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, April 10, 2006 12:51
PM
Subject: [Asterisk-Users]
Hi all,My scenario is this one:LandLine--E1---|-| |---| |OLDPBX|---E1---|Asterisk1.2.5|-VoIPusers
GSMGateway-Analogue-- |-|
> > On Tue, 2006-04-11 at 23:00 +1000, Carey O'Shea wrote:
> >> I only receive 4 google results on my error. So some help would be
> >> appreciated. I could not even determine what "VNAK" was.
> >>
> >> Let me describe my problem. I have an IAX hardware phone here that
> >> connects and operates fi
I think it's a bug. How can we ask the developers to check it?
Cheers
Javier
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Tomislav
Parcina
Enviado el: Jueves, 06 de Abril de 2006 4:37
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto:
On 11 Apr 2006, at 15:33, Pimjai Wesnarat wrote:
Hi,
I still cant dial out on Zap and I really have no clue why.
I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card
4 ports, 31 channels each and able to receive incoming calls and
fax perfectly.
I've done this in my dial p
>> I am still looking for a solution and I am sure that I am not the only
>> one having that problem:
>>
>> If provider A fails for any reason, the next provider should be taken.
> exten => s,n,GotoIf($["${DIALSTATUS}" : "(CHANUNAVAIL|CONGESTION)"]?
> tryiax02:Hangup)
Yes, this is exactly how I'v
On 11 Apr 2006, at 14:52, Carey O'Shea wrote:
Some more info:
Just tried this on a server without using any NAT and no port
forwarding, no masquerading, and I still have the same problem. So
there
goes that idea. I do not know what this VNAK error means.
By the way, I am using the latest v
On 11 Apr 2006, at 14:52, Carey O'Shea wrote:
Some more info:
Just tried this on a server without using any NAT and no port
forwarding, no masquerading, and I still have the same problem. So
there
goes that idea. I do not know what this VNAK error means.
By the way, I am using the latest v
On Mon, 2006-04-10 at 22:42 -0400, Andres wrote:
> Steven wrote:
> You heard wrong. We have multiple PRIs from XO and they DO NOT send
> caller name. We have discussed the issue with them on several
> ocassions. The sales people will say whatever they want, but the tech
> people who actually
On Tue, 2006-04-11 at 15:58 -0400, Mike Clark wrote:
> >That is a positive new :-/
> >Any pointers to a sample? I couldn't find a suitable sample. I don't
> >have much experience with AGI but I can follow a sample if I had one.
> >I usually call a bank's IVR and I'm asked for merchant number, de
Hi everyone:
I installed the lates version of Asterisk with Asterisk Add-Ons. A month ago
i upgraded my database form mysql 4.1 to mysql 5.0. So after to start
Asterisk i have the following error:
[res_config_mysql.so]Apr 11 17:25:51 WARNING[31300]: loader.c:325
__load_resource: /usr/lib/as
Can anyone tell me how me to get asterisk to dial out a phone number
when a bluetooth device is not detected?
I've tried putting the following under the clients section in
/etc/asterisk/btp.conf:
client =>user,00:12:34:56:78:90,Zap/4/1234567890
and in extensions.conf:
exten => 222,1,Playback(pl
>>On 4/10/06, Dov Bigio <[EMAIL PROTECTED]> wrote:
>>
>>Hi,
>>
>>I am using Asterisk for a call center on a Dual Xeon machine..
>>
>>I currently have
>>
>>109 active channels
>>53 active calls
>>
>>Every body is complaining about quality and cpu is around 80% idle.
>>
>>Is there any tuning I can d
We do it slightly different, rather than multiple macros, we do it
within a single macro.
;
;
;
[macro-outbound-calling]
exten => s,1,NoOp("Debug: Outbound Call from ${CALLERID}")
;
exten => s,n(tryiax01),NoOP("Debug [${CONTEXT}]: Trying 1st IAX2
Service")
exten => s
Have you quit and relaunched Asterisk? (not a reload, but a full quit
process and restart) I know in the past when I have a process
already listening to 0.0.0.0 it will not always pick up a newly added
NIC alias address without re-binding.
Daniel
On Apr 11, 2006, at 12:21 PM, Michael Geor
Hi,
I checked core file generated at /tmp after a
downtime, here is what I got...
Is anybody able to interpret what did Asterisk went
down???
Thank you
Dov
Loaded symbols for
/usr/lib/libstdc++.so.5Reading symbols from
/lib/libgcc_s.so.1...done.Loaded symbols for /lib/libgcc_s.so
Would that caching dns daemon be "nscd"? (included in every distro).
I had some problem with it in the past and don´t like it, but it´s
major function is to turn a workstation capable of self-caching DNS
and NIS queries.
andre
On 4/11/06, Joseph Tanner <[EMAIL PROTECTED]> wrote:
> I've had this
Boris Bakchiev wrote:
The simplest solution and the one already implemented in linux, tmpfs.
It would be best to allocate 4-8GB to tmpfs on /tmp and let the kernel
do the work it was designed to do. And you would not be limited to PCI
bus speeds. The DDR2800 is about 12GB/sec. Some would say "ov
Luki wrote:
Has anyone seen these solid state "Drives" from gigabyte yet? -
http://www.pcper.com/article.php?aid=224&type=expert&pid=3
Interesting device. Looks like the burst throughput is right on par
with good drives, but you have better sustained throughput and
obviously near zero lat
I am looking for someone who know what they are doing with a TnT MAX to help me get started with configuring the thing. The unit will have 6 PRI's and 18 E&M T1's going into it and sending the calls out VoIP to asterisk boxes and to upstream voip providers. Has 3 x 8T1 cards and 8 x 96 VoIP DSP car
Matt Roth wrote:
> The last point also brings up a question. Does anyone know how
> gracefully Asterisk handles attempting to write leg files to a full disk?
I suspect it would fail in an ugly way
___
--Bandwidth and Colocation provided by Easynews
thanks!
www.cacti.net -
Open source application for handling SNMP manageable
devices.
There is already a FatPipe host/device template and graph
template that someone has built!
I haven't loaded it as yet-- note his comments about his
RRA being included in his templates.
http://forums.cact
Anyone any ideas?
They are SIP phones. I am not sure if its an asterisk or phone problem. Any
help to isolate would be good.
Thanks
Paul
- Original Message -
From: "Paul A Brown" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, April
> The last point also brings up a question. Does anyone know how
> gracefully Asterisk handles attempting to write leg files to a full disk?
We've had this happen twice and if it is a regular-path partition it
seems to have handled the overlap in either RAM or on the root
partition. Not sure exac
Hello there,
For some reason, Asterisk is crashing everytime that my system tries to do
anything related to ODBC. I would really appreciate if anyone could give me
ideas or pointers to the solution of this issue...
Here's what I've found out so far:
* I can run isql -vwithout any proble
Erick Perez wrote:
How much RAM disk is needed or are you using for your current needs?
We're planning to do something like this. But I can't figure proper
dimensioning.
Erick,
We are using Asterisk to handle our inbound call center operations.
There are currently 158 leg files (produced by
Hi,
Do we need STUN server with Asterisk(1.2.6) for SIP phones which are using
NAT on different networks ???
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
That is a positive new :-/
Any pointers to a sample? I couldn't find a suitable sample. I don't
have much experience with AGI but I can follow a sample if I had one.
I usually call a bank's IVR and I'm asked for merchant number, device
number, etc. The system ask me for credit card number (t
Hi List,
I'm writing a system that issues a lot of automated calls, in an opt-in
basis. I've found that even though the calls are to different
destinations, if they are issued within the same second, they get the
same channel AND the same unique id.
Is there a way to prevent this?
Current
Nobody knows the answer to this!?!?!?
--
~Shaun
>I have a macro that runs off a dial() and gives the callee a bunch of
>options... one of them is to disconnect the caller. I read that setting
>MACRO_RESULT=ABORT would hang up both "legs" of the call. When i set
>MACRO_RESULT=ABORT and retur
Waldo Rubinstein wrote:
> Can anyone provide any further info on External IVR application? It
> seems interesting. I currently have a heavily used AGI script that I use
> for a custom IVR. It is written in Perl. I wonder if it would be more
> efficient to "migrate" it to this External IVR. Will it
Linksys just lost my VoIP business I guess.
> -Original Message-
> From: tracinet [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, April 11, 2006 8:02 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SPA-941/942 Bulk provisioning
>
> Unfortunate
Hi,
I've been battling with a similar issue:
a) I wrote a script to periodically run the command "cat
/proc/interrupts" and figure out the interrupts per second. I run this
script for over 24 hours and never once did the difference between the
preceeding and succeeding interrupt counts go below
Hi niklas,
I know it would have been
span=2,1,0,ccs,hdb3,crc4
But if I try this configuration asterisk zapata seams not to be able to
sincronize.
On logs I have continuosly:
Apr 7 09:03:18 NOTICE[3196] chan_zap.c: PRI got event: Alarm (4) on Primary
D-channel of span 2
Apr 7 09:03:18 WARNING[
I have an * box that I need to chang the IP address on.
My hope was that I could add an alias to the interface with a different
IP address, have * bind to all addresses, change DNS and when no more
hits come on the old address.
However, IAX registrations coming in to the alias don't seem to get
a
On Tue, 11 Apr 2006, Andrew Kohlsmith wrote:
> Please do not open your mouth to spout nonsense if you do not know
> what you're talking about.
[...]
> Again, if the IO-APIC is reporting that the card is on its own IRQ,
> it really, truly, honestly *IS* on its own IRQ. T
Hi Anton,
I'm using a supermicro P4 3GHz P8SCT (Intel E7221 chipset) with TE205P and a
TDM04 and I've similar problem.
I was using linux 2.6.9smp that seams to have problem with APCI so
Hyperthreading, even if enabled, was not working (I sow 1 cpu).
Today I've disable hypertrading and start using m
Does anyone have a SEP.cnf.xml file that works with asterisk? I
have the SIP firmware loaded on my Cisco 7970 but the status log shows
errors parsing the config. I copied a config that was posted to the list
but it didn't seem to work. Any help would be appreciated.
Jeremiah
--
___
I've had this problem too. It would get so bad, that it wouldn't even
answer incoming calls, and if I tried to dial out via pstn, I would
have hung up before it got around to dialing (which it would
eventually do, unfortunately).
A short-short term solution was to install bind, and use it as your
Hi,
Gere are some messages that sometimes show up in my
Asterisk logs... If you help me out to solve them, I could make a list of all
know warning messages so that we can publish in the wiki or somewhere
else!
- "res_features.c: Did not read data." - on Google,
the only reference to this
No, I'm taking receiving CallerID name and *not* sending. and no on a
PRI wait should not be required for callerID to come in.
On 4/11/06, Jerry Jones <[EMAIL PROTECTED]> wrote:
> I CAN VERIFY via aa dozen PRI from XO that yes indeed provide
> incoming callerID on PRI. It arrives shortly after the
Can anyone provide any further info on External IVR application? It
seems interesting. I currently have a heavily used AGI script that I
use for a custom IVR. It is written in Perl. I wonder if it would be
more efficient to "migrate" it to this External IVR. Will it be more
efficient? Will
On 4/11/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
>
> In the US, bri & pri's are less popular for lots of reasons, part of
> which is the cost of implementing the necessary software on the CO
> switch. Siemens (as one example only) charges their small CO customers
> $7,000 to implement the softwa
Hi everybody,
A customer requires G726-40 with Asterisk... I know G726-32 is
pseudo-standard, but he definitely wants G726-40...
Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone
done this before? Any hints? Please help!
Due to a misunderstanding, my product manager already
Hopefully it's okay to *announce* this here.
I've been working on a project for Asterisk for some time and it is
finally ready for a beta release. Any feedback is well appreciated. At
the basic core it's a Dialer for Windows. I'll be adding more features
quickly, but I wanted to keep everything si
On 4/11/06, Andy Tan <[EMAIL PROTECTED]> wrote:
> Hi,
>
> understand that the bandwidth utilized for each call is dependent on the
> codec used, wonder if Asterisk can monitor the total bandwidth utilized
> and restrict/reject new calls when the resource is insufficient to
> support them reliably?
XO CAN supply callerid NAME on a NI2 PRI connection.
We have three of them and they work great. Its takes a little doing to
get to someone at XO that knows what they are doing
but XO does have some VERY good tech support people that know how to
get things done. It just takes a bit of work t
If the agent logs in as an agent thast is a member of the queue, then if
that agent is in multple queues they will only get one call at a time,
regardless or how many queues they are members of. (I hope you were able
to follow that!!)
In regards to the two phones.
1 WHY?!
2 DO both
Title: Re: [Asterisk-Users] Asterisk stops responding when internet is down
Asterisk is sensitive when it comes to DNS lookups. If the DNS server configured on your Asterisk server is not reachable, Asterisk may block while waiting for a result. This can cause chan_sip to hang and not allow phon
On 4/11/06 8:14 AM, "Joao Pereira" <[EMAIL PROTECTED]> wrote:
> Hello to all
> I would like to know some opinions of people that are using billing
> tools for Asterisk.
> Can you please advise me in wich billing tool to I use?
>
> Thanks
> Joao Pereira
> __
On 4/11/06 8:42 AM, "Carsten Bock" <[EMAIL PROTECTED]> wrote:
> Hi everybody,
>
> A customer requires G726-40 with Asterisk... I know G726-32 is
> pseudo-standard, but he definitely wants G726-40...
> Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone
> done this before? Any
"Out of the Box" probably not but with an AGI script this is very
doable:
You can have a script that monitors active calls and the Codecs that are
in use. The script will have to do some math to calculate the bandwidth
in use and then using the variables in Asterisk, Namely SIP_CODEC. If
you are u
Zttool shows no irqmisses on the te110p card?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Kenneth Lussier
|Sent: Tuesday, April 11, 2006 7:05 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] te110p
Hi Andrew...
Thank you very much for the info.
I didn't recompile the kernel, Im using a generic 2.6 kernel but its worth
taking a look at what you said.. Where can I find (which file) the Hz the
kernel was precompiled to?
Also, Im running 1 te110p and 2 tdm cards, probably I'll disable 1 card
+++ Doug Lytle [11/04/06 07:58 -0400]:
> Vikram Rangnekar wrote:
> >Feel free to try it out and send us any feedback you may have.
> >
> >
> Vikram,
>
> A few issues.
>
> 1). Requires to be run on the Asterisk server via Apache. On a
> production machine, I try to keep the services to a min
BTW... another interesting issue.. the phone line (on Verizon's end)
run parrallel to the fence for a good 1/2 mile.
On 4/11/06, Matt <[EMAIL PROTECTED]> wrote:
> Yes sir.. a "fencer" is an "electric fence".We talked to Verizon
> at one point and they really seemed to not care at all.
>
> On 4
Yes sir.. a "fencer" is an "electric fence".We talked to Verizon
at one point and they really seemed to not care at all.
On 4/11/06, Bob McDowell <[EMAIL PROTECTED]> wrote:
>
> Assuming 'fencer' is the same thing as an 'electric fence', this is
> called stray voltage. The farmer should want t
Scratch that :) Figured it out.
On Tue, 11 Apr 2006, Aaron Daniel wrote:
Just doing some test installs of asterisk running on branch (noticed first on
branch), and noticed if you move to virtual terminal 9 (may be different on
everyone else's), the CLI is running. Anyone have any idea how to
because, a this time, the sip stack doesn't have asynchronous DNS... so ALL the sip code is stucked waiting timeouts for DNS queries (that are long timeouts).
When you try to dial a LAN device, the sip code is trying to resolve your voISP service providers' addresses.
We workaround this putting all
Leave it in the ATA! As long as you can live with standardizing on an
ATA. Sipura/Linksys a good choice for VSCs.
Even if you use some IP hardphones as well you will find softkey or
hardkey functions to replace most of the VSCs. (redial, forward, return
call, etc.). IP 501 Polycom has a good featu
Just thought I would post this as someone might find it usefull.
This is the dialplan for making outbound calls from the UK (not
internetional).
It can be set to block callerID for particular extensions. I have also
added some detection of the PRI error numbers when a call fails to give
some extra
Just doing some test installs of asterisk running on branch (noticed first
on branch), and noticed if you move to virtual terminal 9 (may be
different on everyone else's), the CLI is running. Anyone have any idea
how to turn this off?
--
Aaron Daniel
Computer Systems Technician
Sam Houston St
if you have patched asterisk with bristuff, you could use the app DevState(newstate).
Basically, a thing like this:
; suppose 999 is your "nightmode enable/disable" extension
exten => 999,hint,DS/nightmode
exten => 999,1,your enable/disable stuffexten => 999,2,your enable/disable stuff
exten =>
On 4/11/06, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> Kevin P. Fleming wrote:
> > Ronald Wiplinger wrote:
> >
> >
> >> It does not go to the next provider. Is there a settings for "timeout"
> >> to go to the next provider???
> >>
> >
> > Uhh... yeah. That is why there is a timeout parameter for
You could find here an xml example to provisioning them.
http://www.sipura.com/support/spa941faq/index.htm
Kerry Garrison escribió:
Has anyone got any information on bulk provisioning of Linksys
SPA-941/94s? There is an overview in the admin guide but it refers to
a different provisioning guid
Out internet connection was out this morning. It seems that the SIP
extensions on our LAN were affected. Behavior like:
Call comes in over POTS to a TDM400P, there is a delay then before the Cisco
79[46]0's start to ring.
If we were lucky enough to get a call through, then we could not transfer
th
Hi,
understand that the bandwidth utilized for each call is dependent on the
codec used, wonder if Asterisk can monitor the total bandwidth utilized
and restrict/reject new calls when the resource is insufficient to
support them reliably?
Regards
Andy Tan
--
Andy Tan
[EMAIL PROTECTED]
--
h
Hi everybody,
A customer requires G726-40 with Asterisk... I know G726-32 is
pseudo-standard, but he definitely wants G726-40...
Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone
done this before? Any hints? Please help!
Due to a misunderstanding, my product manager alread
maybe this could be solved using "Local" channel as members, and limiting calls to the agent (actually an extension if using Local/[EMAIL PROTECTED]) in the dialplan with GROUP and GROUP_COUNT
2006/4/10, Marco Campos <[EMAIL PROTECTED]>:
Instead of "call-limit=1" try o use "incominglimit=
Hi Tony,
thanks for your answer!
I tryed doing so, but I still get that error, sorry.
Giuseppe
-
Tony Mountifield ha scritto:
In article <[EMAIL PROTECTED]>,
Giuseppe <[EMAIL PROTECTED]> wrote:
Hi,
when I try to use meetme I always hear this error message
"this is not a valid
On Mon, 2006-04-10 at 21:42, Lonnie Abelbeck wrote:
> >
> Adding "defaultip=10.x.x.x" might solve the problem.
>
> GXP-2000's can work without registering, using "host=10.x.x.x" as long as you
> don't want to use BLF with the new firmware.
>
> The new firmware is great, as long as you don't have
1 - 100 of 154 matches
Mail list logo
