Chris Mason (Lists) пишет:
Tzafrir Cohen wrote:
Do you rotate Asterisk's logs with the logger or with logrotate?
I have never addressed this before and never seen this problem before.
The issue is causing thousand of log files to be written to the
/var/log/asterisk directory, so many that I
Hi,
I had similar problems with zaptel on a tdm2400.
I found that with the standard make install, zaptel was being started as a
service but not properly initialising the card.
I disabled the service and added a few bits in rc.local;
rmmod the zaptel modules,
sleep a couple of seconds,
do a
Hi,
After upgrading from 1.2.13 to 1.2.14 it seems that I do not receive any
ExtensionStatusEvents via the manager API anymore. Anyone else
experienceing this? Any thing I need to config?
I diffed 1.2.13- and 1.2.14-versions of manager.c, and found no
differences, so I presume the problem
On 5 Feb 2007, at 21:46, chester c young wrote:
Need to deploy between 50 to 300 lightweight Linux - only browser
and softphone.
You might want to consider our lightweight java softphone (Corraleta
SDK) - it can be embedded in
a web page - zero install/config in the client. The UI is in
Stephan,
Ok, I'll re-state the problem...
I have two devices that I want to talk to each other:
1. an Asterisk PBX
2. a Damm Cellular TETRAFLEX digital radio system (www.damm.dk)
both devices are effectively gateways because they have many subscribers
behind them.
The Damm Cellular system
On Tue, Feb 06, 2007 at 09:03:27AM -, Robert Jenkins wrote:
Hi,
I had similar problems with zaptel on a tdm2400.
I found that with the standard make install, zaptel was being started as a
service but not properly initialising the card.
I disabled the service and added a few bits in
In article [EMAIL PROTECTED],
Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
Steve Davies wrote:
I have a problem understanding which 'h' (hangup) extension is used in
which case - It seems to vary depending on channel type.
It doesn't. It depends on which side of the call hangs up.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Can someone explain what the parameters pridialplan and prilocaldialplan
are? What do they do and do I need them?
I've connected an asterisk box via E1 (sangoma) to an alcatel 4200 pbx.
The pbx technican complains about the format of the nr
Hi,
I was wondering if it is possible to set asterisk in order to listen to
different ports for the sip or I need to do this operation with iptables?
All of this since some time the port 5060 is blocked.
Thank you
___
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Hi,
if anybody interested.problem SOLVED!
The message appears when the user tries to make an outbound call but all
the lines are busy. I tested it only with ISDN (beronet) card, do not
know if happens in other circumstances.
Giorgio Incantalupo
Giorgio Incantalupo wrote:
Hi,
sometimes
**I dont have any problem, my asterisk is working fine. but on the cli,
**asterisk keeps saying Got SIP response 603 Declined (no dialog) back
**from 192.168.0.100. trixbox running on another machine is registered to our
**server from address 192.168.0.100. whats the reason of this msg?
**
**--
I believe it will be hooked up to extension lines.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Monday, February 05, 2007 8:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk
Thanks. Is there a way I can log into the Merlin Magix to determine
that? How else do I tell?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Germann
Sent: Monday, February 05, 2007 8:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial
here's what I found on voip-info.org
http://www.voip-info.org/wiki/index.php?page=Avaya+or+Lucent+Magix+Voice
mail+Integration
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Germann
Sent: Monday, February 05, 2007 8:16 PM
To: 'Asterisk
Andrey Solovjov wrote:
This usually happens if one of the log files in /var/log/asterisk is
more than 2Gb...
I had deleted all the log a week previously, so that's is not likely.
I think we have a bug. I built two systems on the same hardware, the
only difference was one had a Sangoma
Ciao Tony,
I believe that for any channel that is executing in the dialplan,
when it is hung up, either directly or due to its peer hanging up, it
will execute the 'h' extension, if any, in whatever is its current
context.
My experience confirms your statements.
IMHO Asterisk should behave
Hi All,
I want to build Production High density call center using asterisk. May
I please get your view of, which server should I use to deploy my
system?
ZAP TE412P connected direct to Telco via 8 E1
Regards
Sam
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
Patrck schrieb:
On Mon, 2007-02-05 at 11:06 +0100, Tomislav Parčina wrote:
Hi list!
How to make outgoing call thru other mISDN channel group of all channels on
first group are busy?
I believe I'll need to
- Check of there is free channel
Hello Larry,
Probably your variable (MYIP) is not accessible to asterisk process
environment.
Test it with ${ENV(PATH)} and you will have a result there
exten = s,n,Set(test=${ENV(PATH)})
-- Executing Set(IAX2/test_iax,
test=/sbin:/usr/sbin:/bin:/usr/bin:/usr/X11R6/bin) in new stack
--
On Mon, 2007-02-05 at 23:05 -0500, Steve Prior wrote:
Matthew Rubenstein wrote:
The real advantage in choosing an AGI (or CGI or ...) platform/language
is *reusing* the existing code that already runs on that platform, with
Well of course you should pick whatever AGI
Right now I am using analog but the plan is to use PRI for the proof of
concept but the actual system would use PRI. I know that the analog support
is supposed to be somewhat unreliable but I have yet to get it to detect
even a busy - not even once. I can only assume that I missed some setting
Hi,
I'm using Asterisk 1.2.9.1 on a Debian box with a beronet card. I
compiled and installed beronet driver but some callers cannot use IVR on
my PBX because dtmf are not recognized.
I'm going to install junghanns driver to fix this problem. Is there
anybody who experienced bad dtmf detection
Thanks for your reply Ioan.
Very interesting. ${ENV(PATH)} works to display the path
but ${ENV(MYIP)} does not!
There must be a list in Asterisk that only allows cerain environmental
variables to be shown. A very unnecessary bummer.
However, at the CLI prompt:
! echo $PATH and ! echo
On Tuesday 06 February 2007 7:02 am, Andrea Spadaccini wrote:
This would surely be more intuitive and would require less dialplan
programming when there are more than one point where one might get an
hangup and hangup conditions must be handled (i.e.
Vicidial/Astguiclient).
So then just do
hi list
i'm struggling with my new voiceblue gateway an my asterisk...
i've got three simcards from the same provider, all in one voip-gsm-
gateway. how can i tell my asterisk, that the 4th call, to this
provider, can not be routed through the gateway but to another trunk?
asterisk log
Ack. That should be I am using analog for the proof of concept but plan to
use PRI for the actual system.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian K.
Alexander,Jr. (Vision Point Systems)
Sent: Tuesday, February 06, 2007 8:44 AM
To: 'Asterisk Users Mailing
Ciao,
just change port value in sip.conf.
Giorgio
Il Neofita wrote:
Hi,
I was wondering if it is possible to set asterisk in order to listen
to different ports for the sip or I need to do this operation with
iptables?
All of this since some time the port 5060 is blocked.
Thank you
On 2/6/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Steve Davies wrote:
I have a problem understanding which 'h' (hangup) extension is used in
which case - It seems to vary depending on channel type.
It doesn't. It depends on which side of the call hangs up. h is
executed when the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
Sent: 06 February 2007 10:34
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] New user question (X100P)
On Tue, Feb 06, 2007 at 09:03:27AM -, Robert Jenkins
We just had the oddest thing happen which worries us as new users.
We had 3 calls running on asterisk (one from sip to sip and the other to
sip to zap). It seemed for no reason, the connections just dropped and
the lines went dead. You couldn't call a phone (not even yourself). Once
I restarted
Ciao Andrew,
On Tuesday 06 February 2007 7:02 am, Andrea Spadaccini wrote:
This would surely be more intuitive and would require less dialplan
programming when there are more than one point where one might get
an hangup and hangup conditions must be handled (i.e.
Vicidial/Astguiclient).
Thanks for the help!
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
This error message is specifically from running ztcfg.
What do you have in /proc/zaptel/* ? What in /etc/zaptel.conf ?
The difference between the two is the immediate cause for the error.
In /proc/zaptel I
I read all the other replies and want to thank everyone!!
I think most of what other people mentioned is answered below.
I'm only doing this to test Asterisk. I will be using T1 cards when I start
putting Asterisk in production. I've got several Ad-tran TSU 600s and 120s
that I can use for analog
One other note, the output of lspci for the two X100Ps is:
01:07.0 Communication controller: Motorola Wildcard X100P
01:08.0 Communication controller: Motorola Wildcard X100P
Thanks, David
___
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On Tue, Feb 06, 2007 at 10:22:43AM -0500, David Ruggles wrote:
Thanks for the help!
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
This error message is specifically from running ztcfg.
What do you have in /proc/zaptel/* ? What in /etc/zaptel.conf ?
The difference
In article [EMAIL PROTECTED],
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 06 February 2007 7:02 am, Andrea Spadaccini wrote:
This would surely be more intuitive and would require less dialplan
programming when there are more than one point where one might get an
hangup and hangup
What is the output of cat /proc/zaptel/1 ?
(My guess: ztdummy)
Correct
Do you have two cards?
Hmm... you do seem to have two of those...
Either a defective hardware or misunderstandings with the PCI bus.
For instance, on one system I know, I had to pass the kernel parameter
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
Steve Davies wrote:
I have a problem understanding which 'h' (hangup) extension is used in
which case - It seems to vary depending on channel type.
It doesn't. It depends on which side of
On Tue, 6 Feb 2007, Larry Alkoff wrote:
There must be a list in Asterisk that only allows cerain environmental
variables to be shown. A very unnecessary bummer.
There is no list.
Try executing a dialplan containing a priority like:
exten = *,n,system(set /tmp/what-asterisk-sees)
On Tuesday 06 February 2007 11:17 am, Tony Mountifield wrote:
I have never needed explicitly to Goto the 'h' extension. If I'm in a
normal context and a Dial fails, if I then fall off the bottom of the
extension, it goes to my h extension anyway.
I don't think I have to, but I explicitly do
In article [EMAIL PROTECTED],
Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
Steve Davies wrote:
I have a problem understanding which 'h' (hangup) extension is used in
which case -
Since Monday I didn't see much traffic.
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I can confirm,
commands after Wait() are never executed in 'h' extension
and wait seconds argument in wait() is completely ignored
it's bug or feature? ;-)
h = {
NoOP(before ${EXTEN});
Wait(5);
NoOP(after ${EXTEN});
}
-- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/bill-gw-10,
On Mon, 2007-02-05 at 04:46 -0700,
[EMAIL PROTECTED] wrote:
Date: Sun, 04 Feb 2007 23:35:46 -0500
From: Steve Prior [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Which Java FastAGI implementation has
the mostmarket share?
To: Asterisk Users Mailing List - Non-Commercial
Spam detection software, running on the system placebo, has
identified this incoming email as possible spam. The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email. If you have any questions, see
[EMAIL PROTECTED] for details.
Content
Robert Jenkins wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
Sent: 06 February 2007 10:34
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] New user question (X100P)
On Tue, Feb 06, 2007 at 09:03:27AM -,
On Tue, 6 Feb 2007, Steve Edwards wrote:
On Tue, 6 Feb 2007, Larry Alkoff wrote:
There must be a list in Asterisk that only allows cerain environmental
variables to be shown. A very unnecessary bummer.
There is no list.
Try executing a dialplan containing a priority like:
exten =
Hello,
Maybe it is too late but it may help you.
Check the configuration for the SIP client identified by 192.168.0.123
(or the IP mentioned by the error line)because it tries to subscribe
to get BLF indications for the X extension. Most probably it is for
an old phone BLF configuration.
He had it right
He is using Asterisk to REPLACE the Octal, so it needs to be equipped
with FXO
Since most.all VM's should have multiple ports to the PBX, you probably
will want to equip the Asterisk box with a Sangoma card with 2 FXO
modules, for a total of four ports.
This will allow the
On Monday 05 February 2007 5:50 pm, Stefano Corsi wrote:
Ok, understood. But I'm still very curious: what is the wife test ?! :)
This is the test which governs what technology gets to stay in the house
and become used by the family, and what gets banished to the shed at the
bottom of the
I wonder if there are CDs available for purchase. I don't have any way
to burn one from a downloaded iso image. Any help appreciated.
Tom
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To UNSUBSCRIBE or update
On Tue, Feb 06, 2007 at 08:06:42PM +0200, yusuf wrote:
I have seen this also, only on the TDM2400. I think it might be because
it, i.e. this cards, takes a bit longer than other cards to initialise,
then when ztcfg is run, the card is not ready yet.
So I too (hangs head in shame), put
How do you start Asterisk? You need to make sure the environment
variable you want inside Asterisk is being exported. I use 'export
HOSTNAME' in my asterisk init script and it works like a charm.
Larry Alkoff wrote:
Thanks for your reply Ioan.
Very interesting. ${ENV(PATH)} works to
Stefano Corsi wrote:
Eric \ManxPower\ Wieling wrote:
There's still something I don't understand: when using a simple modem
on an analog line, you get correct answers from the modem: NO
ANSWER, BUSY, NO DIALTONE, etc... why is this possible with
these TDM2400 cards that cost twenty times as
On Tue, Feb 06, 2007 at 01:36:58PM -0600, Tom Poe wrote:
I wonder if there are CDs available for purchase. I don't have any way
to burn one from a downloaded iso image. Any help appreciated.
Get any Linux distribution. You can purchace a CD in just about
anywhere. CentOS (http://centos.org)
On Tue, Feb 06, 2007 at 08:04:23AM -0600, Larry Alkoff wrote:
Thanks for your reply Ioan.
Very interesting. ${ENV(PATH)} works to display the path
but ${ENV(MYIP)} does not!
There must be a list in Asterisk that only allows cerain environmental
variables to be shown. A very unnecessary
Thanks everyone!
I removed the extra X100P and tried the remaining X100P in both PCI slots
and it works in one and doesn't work in the other.
I really only need one for early testing so this is good.
Thanks again!
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data,
We have it working fine on an SPA-3000.
CP
On Feb 5, 2007, at 10:42 PM, Joseph wrote:
I've a problem with inserting a pause and dialing additional numbers
when going through Sipura-3000
exten = _12,1,Dial(SIP/[EMAIL PROTECTED],30,D(ww18))
D() doesn't work as it sends the DTMF tones
The best bet, however is to invest the $29 for a CD Burner (or better,
$39 for a DVD Burner) so you can continue to make your own. It's a
cheap investment with a great payoff in terms of burning your own
operating system CDs, making backups, etc.
Tzafrir Cohen wrote:
On Tue, Feb 06, 2007 at
Pavel Jezek wrote:
I can confirm,
commands after Wait() are never executed in 'h' extension
and wait seconds argument in wait() is completely ignored
it's bug or feature? ;-)
h = {
NoOP(before ${EXTEN});
Wait(5);
NoOP(after ${EXTEN});
}
*snipped
in ael use WaitExten(5);
I have a server which I haven't installed that I have to maintain. This server
uses MySQL, it has an asterisk database and in there some mysterious tables:
stats_action, stats_agent, stats_callid, stats_config, stats_estados,
stats_qstats, stats_queue, stats_queuexagent. I say mysterious
Hi list,
I'm getting an error on a E1 link to the telco, after some weeks of
operation without trouble.
I have an asterisk with a TE405 in passtrough mode: two E1 are connected
to the Telco, two E1 are connected to 2 Siemens PaBX. Only 15 channels
are used on each E1 (conf is attached).The
Now that ztcfg is working correctly I can't seem to get asterisk to answer a
call.
I did the make install and make samples so I have a lot of configuration
files that I know nothing about.
Here is contents of zapata.conf
[trunkgroups]
[channels]
context=incoming
signalling=fxs_ks
Tzafrir Cohen wrote:
On Tue, Feb 06, 2007 at 01:36:58PM -0600, Tom Poe wrote:
I wonder if there are CDs available for purchase. I don't have any way
to burn one from a downloaded iso image. Any help appreciated.
Get any Linux distribution. You can purchace a CD in just about
Yes, but you still can download the iso and burn it by yourself. Speed
repeateability are going to make that cheaper reliable than get it by
mail.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Poe
Sent: Tuesday, February 06, 2007 3:48 PM
To:
I'm missing chan_zap.so, I'm going to make and make install again as per:
http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL
I'm in the middle of trying to solve that very problem -- integrate a legacy
pbx into my asterisk/IP network. (Norstar MICS for the record)
Been mulling over how to do it with Quintum boxes the last few days --- very
complicated. The alternative solution is to use a Citel gateway with the
30+
WaitExten is useless in this case, because it's waits for user input,
but we are talking about executing diaplan when entering 'h' extension,
ie. after user hangs up phone...
and seems, something strange with processing wait() app in processiong
'h' extension in diaplan - timeout specified is
Well that didn't work. I still don't have a zap channel driver. What else
can I try?
TIA!
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi list,
I'm getting an error on a E1 link to the telco, after some weeks of
operation without trouble.
I have an asterisk with a TE405 in passtrough mode: two E1 are connected
to the Telco, two E1 are connected to 2 Siemens PaBX. Only 15 channels
are used on each E1 (conf is attached).The
Can anyone confirm that it actually works in Singapore with Busy Detect?
I have a system with loopstart and BusyDetect and have recently attempted to
improve disconnect detection results with the addition of
hanguponpolarityswitch ... results are mixed
--
Chris Earle
System Solutions Specialist
On Tue, 2007-02-06 at 12:21 -0800, Anthony Rodgers wrote:
We have it working fine on an SPA-3000.
Can you post that line of your extension.conf ?
--
#Joseph
CP
On Feb 5, 2007, at 10:42 PM, Joseph wrote:
I've a problem with inserting a pause and dialing additional numbers
when
Let's see if I remember this, it gave me a bit of trouble as well.
*after* you made sure you've got the zaptel driver in order, go to the
src folder for asterysk and issue make menuconfig.
Go to 3 and see if you have the chan_zap listed there and with [*]
prefix. If it's not listed it's because
Thanks for the reply, but when I go to the asterisk source directory and
issue make menuconfig I get:
make: *** No rule to make target `menuconfig'. Stop.
The source I have is the latest tar file from the astrisk site.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe
TP'n to follow flow
waiting for user input that will never come is the same as just waiting G
as for the 'h' aspect to this i must have missed that part.
why aren't you using DeadAGI?
Pavel Jezek wrote:
WaitExten is useless in this case, because it's waits for user input,
but we are talking
I'v got Asterisk 1.4.0 and it understands make menuconfig. Is your
version older or newer? If it's older, maybe you can try the newer one.
If it's newer - I'm out of ideas.
David Ruggles wrote:
Thanks for the reply, but when I go to the asterisk source directory and
issue make menuconfig I
Sorry about that I must have been in the wrong directory. I also have 1.4.0
and I tried it again and it worked. Chan_zap is not listed there, I'll start
poking around and see if I stumble across anything. Do you know where the
expected location is? I don't have a problem moving the source.
Try it like this:
cd /usr/src/asterisk-1.4.0
make clean
./configure --with-zaptel=/usr/src/zaptel-1.4
make menuconfig
make all
make install
David Ruggles wrote:
Sorry about that I must have been in the wrong directory. I also have 1.4.0
and I tried it again and it worked. Chan_zap is not
We have done limited testing with the Citel gateways and they seem
pretty cool. We're fixing to deploy them as a replacement to a hotel
pbx, and after that use them as an interim solution until full VoIP
convergence in our campus environment. I would be interested to know
what other peoples
All,
I'm running (attempting to) ooh323 with Asterisk and a Cisco 2621XM router
operating as a H.323 GateKeeper, however when I bring the Asterisk box up it
registers successfully with the GateKeeper (exchanges GRQ/GCF, then RRQ/RCF) it
notes the GateKeeper supports keepalive at 300 seconds,
I'm struggling to get my VOIP installation to be acceptable. I'm
looking for advice on what else I can look for.
My system:
o Teliax VOIP service, voip-ny1 proxy
o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms
average jitter)
o 3.2 GHZ P4 Server (runs asterisk, firewall,
Run mtr on your server against the registration server at Teliax and
look for bad hops on your route to and fro.
If you don't find anything there, you may want to fire up ethereal
and capture packets on a few calls and look through them for error
data that may be contributing to bad voice
Jim Duda wrote:
I've been on the shorewall firewall and confirmed that I have the
firewall configured properly for VOIP QOS.
What exactly have you done here? You do know that you are apparently
using IAX2 and not SIP. Those are not the same protocols. In fact, if
you configured the
I could have sworn I saw a post about this recently but I can't find it
so apologies if this is a dupe, but is there anyway to control the order
in the Polycom Buddies list?
Bill
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Hi, all, Do you guys happen to know that there are any IP phones have
such feature, that it can has some indication for the agent status
linked to the phone? E.g some LED show the status, backend we can link
the phone to one agent id, then the agent login the system, the 'online'
indication
Assuming you are using a central provisioning server, check your
{MAC}-directory.xml file. It contains the ordering that you are
looking for.
I hope this helps.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
Thanks for the reply Lacy.
Yes, I know that I am using IAX2 and not SIP for my connection to
teliax. IAX2 is the preferred protocol for connection to teliax. I
have the firewall configured to prioritorize port 4569 for IAX2.
I have the shorewall tcdevices file setup with 3 mbit download
After reading through several recent threads, I started to wonder why the
Cisco document (and other VoIP documents) appears to present this issue as
VoIP gateway specific. Don't (plain old) PBX' face the same issue if they
use analogue interfaces? If there are analogue PBX' at all, how do
I was only trying to demonstrate that my special variable MYIP was
indeed in the environment of the shell. I suspect it's not in the
Asterisk process environment - why I dunno.
I'll look at that tomorrow but suspect I'll never be able to read the
MYIP variable from Asterisk.
Larry
It's pretty easy to validate NANP #'s. Is there any practical way to
validate an international number? I know the quick answer is no because
each country manages its own plan. But has anyone tried to compile a list
of plan patterns, country specific if necessary? Has anyone tried to do
Yuan LIU wrote:
After reading through several recent threads, I started to wonder why
the Cisco document (and other VoIP documents) appears to present this
issue as VoIP gateway specific. Don't (plain old) PBX' face the same
issue if they use analogue interfaces? If there are analogue PBX'
From: Larry Alkoff [EMAIL PROTECTED]
I was only trying to demonstrate that my special variable MYIP was indeed
in the environment of the shell. I suspect it's not in the Asterisk
process environment - why I dunno.
I'll look at that tomorrow but suspect I'll never be able to read the MYIP
From: Gordon Henderson [EMAIL PROTECTED]
I define
[globals]
myvar = ${DB(store/myvar)}
---
But when I want to use ${myvar} in the dial plan, I found that the
variable is null when Asterisk is restarted. Only a reload would force
globals to read AstDB. Other variables in globals loads
${DIALSTATUS} is set only after hangup, try adding the NoOp() line to your
'h' (hangup) extension like this:
exten = h,1,NoOp(${DIALSTATUS})
...
This is how I get the status of the call attempt, whether it's done through
Originate or just plain manual dialing.
On 2/1/07, Michael Collins [EMAIL
Helo,
I have problem with Digium TE110P connected to CISCO 3640 (port on
NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I
tried with real PBX problem was exactly same.
I have this messages in Asterisk conole and log sometimes:
NOTICE[1115] chan_zap.c: PRI got event: HDLC
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial attempts in
the s extension. Goto() is used in examples. Is the prefix s- mandatory?
Is it related to the original extension s? (Apparently Goto(${DIALSTATUS})
won't work for me.)
Yuan Liu
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