On Wed, Oct 12, 2005 at 01:44:34AM -0400, Andy Goss wrote:
> If anyone could tell me what this error is all about, I would be very
> grateful.
>
> Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock
> path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not
> pe
On Tue, Oct 11, 2005 at 10:09:34PM -0400, Andy Goss wrote:
> Update -
>
> I made a backup of my entire voicemail directory then deleted it. If I
> then try and record a greeting, it works. Asterisk creates the folder
> structure and records the greeting. If I try to copy the old file back
> in
On Tue, Oct 11, 2005 at 05:37:12PM -0600, Ryan Hulsker wrote:
>
> mine looks like this
>
> #!/usr/bin/perl
> # Takes 3 command line args, context, mailbox, password
> # updates the mailbox password in mysql
>
> use strict;
> use DBI;
>
> my ($Context, $MailBox, $Password) = @ARGV;
>
> my $dbh
On 10/12/05 13:00 trixter http://www.0xdecafbad.com said the following:
Where I got the data from and all is also on that page if anyone wanted
to make their own lists. I would appreciate any updates or corrections
that anyone happens to notice.
a simple modification which would make this a
If anyone could tell me what this error is all about, I would be very
grateful.
Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock
path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not
permitted
Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed
Hey Cory!
How many PCI does it support? Im looking for an option that can support 3 or
more TDM cards, the idea I have is to have model #'s for a server that can
handle 1 E1/T1 cards nicely and one # for a server that can handle 3 or more
TDM cards (without IRQ conflicts) so I can offer both TDM o
Update -
After we converted all the audio to gsm, it miraculously started
working. I still don't know why. If anyone knows how the wav49 codec
or whatever can get screwed up, your input is still welcome.
Thanks and goodnight,
Andy
--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 5
I spent the last day or so gathering every country plan and listing
prefixes as mobile, premium, etc. If anyone wants this I have made it
available at http://www.0xdecafbad.com/Global-Numbering-Plan.html
Each country has its own context, making it easy to include what you
want where. Obviously t
AK - this is a nice configuration for use with Asterisk. If you want
cheap, there are cheaper alternatives, but this configuration works
flawlessly in our experience and is affordable. The SuperMicro server
model # I am referencing is SYS-5013C-MTB and it is a current model, but
it's barebone
See this link: http://www.voip-info.org/tiki-index.php?page=NFAS
- Original Message -
From: "Billy Huddleston" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, October 12, 2005 12:57 AM
Subject: Re: [Asterisk-Users] Dual PRI fail over
NFAS isn't needed for redundancy.. You can do this with 2 PRI's just have
them in the same huntgroup at the CO.. All NFAS really does is free up a
extra B Channel.. For 2 PRI's it makes no sense.. For anything over 2 you
pickup 1 extra B channel per PRI.. We use NFAS on our RAS setup for the
I
Tom wrote:
I currently have a single PRI however we are getting a second PRI, and the
provider (qwest) wants to know if our PBX supports GSAS (they say its a
redundant d-channel technology but searching on google for GSAS reveals less
than nothing). I've set something similar up before on a cis
Guys.
Anybody using supermicro mobos and chassis with TDM cards?
I would like to know which models are you using (mobos and chassis and also
CPUs) and how many TDM cards have you been able to put in without having IRQ
issues like in other cases.
Ive read supermicro servers play nice with asterisk
you should also use leavewhenempty=yes
as the next priority in extensions.conf to goto voicemail ,
the feature works like "when no agent is logged it will go to the voicemail directly".
Ben merrills <[EMAIL PROTECTED]> wrote:
In your queue entry, add the linecontext = voicemail_contextmake 0 in th
If you have a PRI, many vendors will support sending calls to an alternate
destination if the T1 is down. SBC, for example, calls this "enhanced alternte
routing." If the T1 fails, call are routed to the destination of your choice at
the SS7 switch.
Paul
[EMAIL PROTECTED]
--- Tom <[EMAIL PROTECT
I currently have a single PRI however we are getting a second PRI, and the
provider (qwest) wants to know if our PBX supports GSAS (they say its a
redundant d-channel technology but searching on google for GSAS reveals less
than nothing). I've set something similar up before on a cisco 5350, whe
Hi,
Do cards like TE110P and modules X100M and S100M have their serial numbers
(SKD) recorded into the device? If so, how could I extract this info?
TIA,
--hg
___
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Asterisk-Users mailing
Or explained more clearly
The fallback rule is "n + 101", so your voicemail busy priority needs to be 103
(2 + 101).
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600
On Tue, 2005-10-11 at 19:14 -0700, Jesse Keating wrote:
> When I hang up in voicemail, Asterisk seems to stop responding. (hangup
> vs pressing # to disconnect). After that, no calls can be made until I
> restart Asterisk. In IRC, a developer seemed to think it had to do with
> me using switch =
When I hang up in voicemail, Asterisk seems to stop responding. (hangup
vs pressing # to disconnect). After that, no calls can be made until I
restart Asterisk. In IRC, a developer seemed to think it had to do with
me using switch => in my dial plan. Basically I never see the calling
extension
Update -
I made a backup of my entire voicemail directory then deleted it. If I
then try and record a greeting, it works. Asterisk creates the folder
structure and records the greeting. If I try to copy the old file back
into the directory, it wont work. It's the same file name and
everything
My preferred behaviour would be to completely erase the clid from the channel
after the call.
Obviously, this would only be done for zap channels configured with their clid
set to "asreceived".
It would also only be desirable after any deadAgi scripts had run, or even by
using a command as p
Hi
I have gone thru the steps of installing
AreskiCC, I would like to know how to get access to the GUI interface of this
application.
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Asterisk-Users mailing list
Ast
What you are saying is very interesting as well as important, I will
investigate this.
On 10/11/05, Rod Bacon <[EMAIL PROTECTED]> wrote:
> That I can live with.
>
> The main issue I have is that A calls B through a bridged zaptel call. The
> CLID
> on channels used by this call both show A's CLID
Priority 103 should be dial and not hangup, that way it will do the
voicemail stuff with the wait as well.
On 10/11/05, Zack Odell <[EMAIL PROTECTED]> wrote:
>
> Greetings,
>
> I have implemented the following command to allow CNAM to be delivered to my
> users.
>
> exten => 9969,1,Wait(1)
>
> Thi
Yes, the path exists, the files exist, and the permissions all are 755,
all owned by root and in group root. I cant figure it out for the life
of me.
--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]
> -Ori
Andy Goss wrote:
Oct 11 20:32:58 WARNING[3048]: file.c:415 ast_filehelper: Unable to open
fd on /var/spool/asterisk/voicemail/default/5926/INBOX/msg.wav
Oct 11 20:32:58 WARNING[3048]: file.c:793 ast_streamfile: Unable to open
/var/spool/asterisk/voicemail/default/5926/INBOX/msg (format u
I can not figure out what the heck is going on. I went back to my old
version and I still get errors when the voicemail system tries to load
any of the greetings, unavail messages, etc. the normal voicemail
prompts work, but any user recording don't work. Leaving a new message
appears to work, b
Greetings,
I have implemented the following command to allow
CNAM to be delivered to my users.
exten => 9969,1,Wait(1)
This works great!
However it has spawned a new problem. When
this is implemented into a full dial plan. If a user is set to DND or
sends a call to Voicemail by hitt
I migrated to a new version of the voip connection vs1 server software
and I am now getting these errors when I try to call my voicemail. Any
thoughts? The files are there, so I don't get it.
Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav
file 49
Oct 11 19:57:26 WARNING[
That I can live with.
The main issue I have is that A calls B through a bridged zaptel call. The CLID
on channels used by this call both show A's CLID - no problem.
This call ends, and our system receives a test call from C on one of the
channels used in the 1st call. Even though our applicat
Juan,
I just went through the same thing here. I never did get it to work, i suspect it is a bug but have not had time to look into it.
What I did was use the externpass configuration directive in voicemail.conf to run an external perl script which updates my database.
externpass will run y
Yeah, meaning that when A calls B with callerID of 123 and B transfers
A to C and while doing that the callerID is changed to 456 (callerID
from B) then the CDR will show 456 is src.
If you are trying to do billing based on this info then you are out of
luck, as this is not accurate, rather look at
The wiki is located at:
http://www.voip-info.org/wiki-asterisk
Google is located at:
http://www.google.com
and the list archive is located at:
http://lists.digium.com
On 10/11/05, Rod Bacon <[EMAIL PROTECTED]> wrote:
> If you want my opinion, a single server (or even a small farm) is still easir
If you want my opinion, a single server (or even a small farm) is still easir to
manage with conf files.
A simple reload in the * CLI, and you're done.
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia.
I'd be interested to know if this gets worse over time.
Shutdown asterisk, remove card driver, load card driver, load asterisk then
test.
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phon
Bob Goddard wrote:
On Tuesday 11 Oct 2005 22:41, Lee Howard wrote:
Tom Rymes wrote:
Use the right tool for the job!!!
Use a hardware based DSP for faxing not software based.
Why is a soft-DSP to be considered any less-capable than hardware ones?
The reason why I put IAXmo
I posted something on this a week ago, at which time I was told that this was an
'old' issue. Since then, I've spent hours looking, but can't find the answer.
For some reason, some of my CDRs (both to CSV and MySQL) are being written with
the wrong callerid. As best as I can determine, they are
The hardware echo can will probably add a fair bit to the cost. If a person
could add it later on only if they need it that would be a GREAT feature
IMHO.
I also recall hearing that the FXO or FXS modules will be two port modules
(ie. 2fxo or 2fxs).
> -Original Message-
> From: Cory Andr
On Tuesday 11 Oct 2005 22:41, Lee Howard wrote:
> Tom Rymes wrote:
> > Frankly, I would recommend that you forget about trying to fax with
> > Asterisk. Buy a good Multitech analog modem and install HylaFAX.
> >
> > Use the right tool for the job!!!
>
> Actually, you can use HylaFAX and Asterisk to
I think there is a negligible difference btw 14844 and 14500 so it was
rounded down to 14500. When you do the test you will understand why.
I am still not sure if those numbers are correct though as they are full
scale so that would make them 0dbm? To achieve -3dbm I believe you would
want to s
I am having echo problems as well. One thing a Cisco guy recommended to
me was adjust the txgain down, and see if the echo gets better. I
changed mine to -10, and that drastically improved the echo. What the
guy was explaining to me was that there could be a device on the line
someplace that is inc
Hello all. I
am new to Asterisk as well as this group so please excuse me for a bit as I
learn the
ropes of
Asterisk. Anyway, I currently am using a pap2-na adapter with Teliax
and Mesa Networks (my isp) and
was wondering what I
will need to get Asterisk running correctly. I am wonderi
Andrew Kohlsmith wrote:
On Monday 10 October 2005 08:12, Federico Alves wrote:
reverse code and it surely is a legitimate operation. Open source is far
more convenient, but how do we charge for the product? The business model
is not there: the more popular the product is, the more remote the
po
Tom Rymes wrote:
Frankly, I would recommend that you forget about trying to fax with
Asterisk. Buy a good Multitech analog modem and install HylaFAX.
Use the right tool for the job!!!
Actually, you can use HylaFAX and Asterisk together.
https://sourceforge.net/projects/iaxmodem/
Just b
I'm running CVS Head on OS X server. I've had zero luck getting
realtime to install the necessary components. Is there anyone on
here who is looking to do the same and can help me on/off the list?
Thanks! - HJ
___
--Bandwidth and Colocation spo
On Tue, 2005-10-11 at 17:01 -0400, Tom Rymes wrote:
> Frankly, I would recommend that you forget about trying to fax with
> Asterisk. Buy a good Multitech analog modem and install HylaFAX.
>
> Use the right tool for the job!!!
Asterisk may be able to fax better in the somewhat near future. One
On Tue, 2005-10-11 at 19:54 +0100, Bob Goddard wrote:
> On Tuesday 11 Oct 2005 12:19, trixter http://www.0xdecafbad.com wrote:
> [...]
> > It started with the UK as an *example* and everyone seems to have
> > latched onto that. I wanted to know more than the UK, I wanted every
> > country. astbil
Frankly, I would recommend that you forget about trying to fax with
Asterisk. Buy a good Multitech analog modem and install HylaFAX.
Use the right tool for the job!!!
Tom
On Oct 11, 2005, at 10:32 AM, JC van der Walt wrote:
Hi All,
- I have Digium cards and given that the archives point ou
Hello.
One question...
When we use voicemail with flat file configuration (voicemail.conf)
the vaicemail user can change his password by voicemailmain (voice menu)
this change the value in voicemail.conf.
When we use Realtime the password is stored in the database. What the
voicemailmain (voice
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Andrew Kohlsmith
> Sent: Tuesday, October 11, 2005 9:31 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] PRI echo issues: solvable?
>
>
> On Tuesday 11 October 2005 11:49, alan wro
On Tue, Oct 11, 2005 at 10:19:29AM -0400, Paul wrote:
> Kanuri, Seshu (Company IT) wrote:
>
> >You may also like to check the Asterisk management module developed for
> >Drupal at http://www.drupal.org
> >
> This looks interesting. I already have a drupal installation so I will
> find some time
I think the features you're looking for are:
[featuremap]
blindxfer => ??
atxfer => ??
For example, I use ## for blind and ** for attended. This lets me dial
* and # normally, and asterisk only intercepts them if pressed twice in
quick succession... so you could put the T back in the Dial sta
Yes, but you don't need the ${EXTEN}:
exten => 18004267378,1,Answer
exten => 18004267378,2,Dial(Zap/g1/18004267378)
exten => 18004267378,3,Congestion
Andy Goss wrote:
Watch the output of 'pri debug span 1' on the Asterisk server while
placing the call - bug #4468 (http://bugs.digium.com/view.php
Thanks for your kind reply...
The problem is solved: without the "T" option now the '*' button is just an
another DTMF tone... ;D
By the way, the features.conf file is empty, and I'd like to know how to set
or to change the code for forward a call (*21, i.e.)
> Watch the output of 'pri debug span 1' on the Asterisk server while
> placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468)
> might be relevant.
Yes, this is exactly what is happening. Thanks a lot. I am thinking about
adding a special case for the IBM 800 number since it is
On Tuesday 11 Oct 2005 12:19, trixter http://www.0xdecafbad.com wrote:
[...]
> It started with the UK as an *example* and everyone seems to have
> latched onto that. I wanted to know more than the UK, I wanted every
> country. astbill seems to have that data, I seem to have located all
> the litt
Watch the output of 'pri debug span 1' on the Asterisk server while placing the
call - bug #4468 (http://bugs.digium.com/view.php?id=4468) might be relevant.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Andy Goss
> Sent: Monday, October 10, 2005 5:5
That description is correct in my recollection, I received a
presentation on the product a few weeks back and you do daisy chain the
cards together while only technically occupying (1) PCI slot.
Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
45
> I saw an add in my latest Linux Journal advertising Sangoma's new "AA
> series" of FXO/FXS analog cards with on-board echo cancellation, but I
> can't find any information at all on them. Even the link given in the
> advertisement is a dead end as far as I can tell. Anybody else
> seen/heard anyt
Is there a way I can tell if it is asterisk or the carrier that is
timing out from the CLI?
Sorry, I don't have PRI and don't know the details.
However, I'm sure that if you set a high enough verbose or debug
level, you'll see the ISDN messages between * and the carrier's
switch. I don't know w
On Monday 10 October 2005 08:12, Federico Alves wrote:
> reverse code and it surely is a legitimate operation. Open source is far
> more convenient, but how do we charge for the product? The business model
> is not there: the more popular the product is, the more remote the
> possibility of the cre
Yep & thanks for the reply,
I figured out pretty quickly after one test that the /s did not work.
The issue remains that I have been unsuccessful in getting an incoming
call to come into any other context other than the one specified in
sip.conf [general] section
Anything I'm missing here?
I hav
Tzafrir,
The problem was indeed as you discribed. Digium support
help me to straight out my setting by doing exactly what
you said. I was simply mixed up with channels config
by trying different combination of signal, channels value
in zaptel.conf and zapata.conf.
Thanks a lot,
Min
> -O
--- "Kevin P. Fleming" <[EMAIL PROTECTED]> wrote:
> It _IS_ true, the code that does that work was even
> recently improved.
Keven,
True, my mistake. I had 2 asterisk boxes, one running
ver 1.0.7 and the other running 1.2.0-beta1.
The log I posted before was from the 1.0.7, when I
looked at the lo
Seth - we are supposed to have samples of the new analog Sangoma product
in our booth at the upcoming Internet Telephony show in LA. Sangoma is
using a universal PCB board for both digital and analog boards, and you
can take an unpopulated PCB and add FXS/FXO and a plug-in echo can
module as w
> Hello All,
>
> I saw an add in my latest Linux Journal advertising Sangoma's new "AA
> series" of FXO/FXS analog cards with on-board echo cancellation, but I
> can't find any information at all on them. Even the link given in the
> advertisement is a dead end as far as I can tell. Anybody else
Hello All,
I saw an add in my latest Linux Journal advertising Sangoma's new "AA
series" of FXO/FXS analog cards with on-board echo cancellation, but I
can't find any information at all on them. Even the link given in the
advertisement is a dead end as far as I can tell. Anybody else
seen/heard an
http://voip-info.org/wiki/ is your friend.
More specifically:
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime
HTH
Steve
P.S.
Google should be your best friend, always ask him questions before the
mailing list ;-)
- Original Message -
From: julien bos
To: asterisk-u
http://www.voip-info.org/wiki/view/CID+Issues+with+some+Siemens+DECT+phones+in+France
OMG I posted something useful
But it was someone on the list here who gave me that solution
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Asterisk-U
I'm not sure any more but; I think like 2400 per day. You can tell by just
putting a number in it won't accept a value over it's limit.
Unfortunately it isn't real high.
--
From: Geoff Manning[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussi
On Tue, 2005-10-11 at 11:16 -0500, Kevin Scott wrote:
> I'm curious, for a $2511/min call, which +1 number was this? +1900?
>
> Kevin
809. It was set up as a premium number in that country. While that was
an extreme case it did aparently happen back in the 90s sometime. And
because the country
On 08:51, Tue 11 Oct 05, Jordan Bean wrote:
> exten => 400,1,Dial(SCCP/401&SCCP/402&SCCP/403&SCCP/404,20)
Maybe you can try something like:
Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED])
That works great in our setup
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG
On Tuesday 11 October 2005 11:49, alan wrote:
> After solving the other "low hanging fruit" audio issues in our Asterisk
> PBX, we are left with occasional cases of severe echo which we have not
> found a solution for yet.
> Our system:
> - Asterisk 1.2.0-beta1
> - TE110P on a PRI
> - TDM04 and TD
On Tue, 2005-10-11 at 16:47 +0100, Steve Kennedy wrote:
> > And outside the UK you may want to know what is premium, mobile, etc.
> > What would incur higher charges for the call itself.
>
> You can find that out from the Ofcom number plans
They have lists for outside the UK? I didnt think they
Eric "ManxPower" Wieling wrote:
>>
>>> span=1,1,0,d4,ami
>>> e&m=1-24
>>>
>
> Looks like you have told Asterisk to get it's timing from the Mitel.
> I'll bet the Mitel is trying to get it's timing from Asterisk.
>
> Try span=1,0,0,d4,ami and run ztcfg -vvv
We turned on the zaptel debugging an
I'm curious, for a $2511/min call, which +1 number was this? +1900?
Kevin
-Original Message-
You think country code 44 is a mess, think about country code 1, it
spans many countries ... some in +1 have had $2511/minute rates. Yes
twenty five hundred eleven united states dollars per min
> Subject: Re: [Asterisk-Users] SPA-841 "Decode Latency"?
"Matias G." <[EMAIL PROTECTED]> wrote:
> > Luki <[EMAIL PROTECTED]> wrote:
> >
> >> > Does anyone have any familiarity with "decode latency," specifically
> >> > with Sipura devices? Why would it take 200+ms to decode a 20ms RTP
> >> > pac
On Tue, Oct 11, 2005 at 02:21:03AM -0700, trixter http://www.0xdecafbad.com
wrote:
[snip]
> > Also you have to know who's terminating it. You can make an assumption
> > re BT termination, but directly connected businesses may use another
> > telco with different termination rates etc.
> > A lot o
Hello,
After solving the other "low hanging fruit" audio issues in our Asterisk
PBX, we are left with occasional cases of severe echo which we have not
found a solution for yet.
Our system:
- Asterisk 1.2.0-beta1
- TE110P on a PRI
- TDM04 and TDM40, but these are unrelated to current echo issues
Hi!
Perhaps newbie but I cant find somewhere to set the
TTL for sip registration when * acts as client
Regards
Anders Svensson
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Aster
On Tue, Oct 11, 2005 at 02:29:04AM -0700, trixter http://www.0xdecafbad.com
wrote:
> Yeah but that still doesnt answer the fundamental question. While they
> do have who owns stuff they do it based on prefix (ie 44 871 59 is
> pipemedia) but it doesnt tell you how many digits are past that (5 mo
I am having some difficulties with fxotune on asterisk 1.20beta1. I
followed the information on the wiki about running fxotune and made sure to
stop asterisk, etc. When running fxotune -i 4 I get the following error
message
Could not fill the input buffer
Tuning module 1.failure
Could not fi
Thanks Olivier
Actually, today I have tried the patch provided by
[http://www.lusyn.com/asterisk/patches.html]
as indicated on the Asterisk and UK Caller ID page
[http://www.voip-info.org/wiki/view/Asterisk+and+UK+Caller+ID]
It works now !
My card is a X100P clone, (Tiger3XX Modem/ISDN)
The
On Tue, 2005-10-11 at 17:28 +0200, Olivier Perrin wrote:
> Hi Guillaume,
> You can find a solution here :
> http://www.voip-info.org/wiki/view/CID+Issues+with+some+Siemens+DECT+phones+in+France
> Was working for me few month ago.
> Regards
> Olivier
Thanks Olivier, just got my callerid back on a
Dennis Walker wrote:
>
> But I did find that down in the t1 parameter settings you can set the
> limits higher. I maxed them out and the problem went away, it would
> reset the count in a rolling 24 hours luckily the slip count just
> stayed below the limit.
>
Do you by chance now what the max
Cool, now if only it was available in E-1, and certified for use in
Australia. Actually this is pretty much what I was thinking of building
myself :) Now I know it can be done. Yippee!
Craig
- Original Message -
From: "astgroups" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List
Hi Guillaume,
You can find a solution here :
http://www.voip-info.org/wiki/view/CID+Issues+with+some+Siemens+DECT+phones+in+France
Was working for me few month ago.
Regards
Olivier
Le lun 10/10/2005 à 19:28, guillaume a écrit :
> Hi all
>
> I try to get the caller id of a incoming call through a
On Tue, 2005-10-11 at 17:06 +0200, Joseph Rothstein wrote:
> Thanks for the reply Bret.
>
> I have tested this parsing issue ever way possible, equal variables unequal
> variables, arithmetic, ie $[1 + 1], etc., etc., and my conclusion is that
> Solaris does not parse the $[expr1 operator expr2]
Crystal Stream, Incorporated wrote:
I'm getting:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type:
IAX Subclass: REJECT
Timestamp: 9ms SCall: 8 DCall: 2
[66.98.180.77:4569]
CAUSE : No authority found
Under the DIDx number I'm putting IAX:
[EMAIL PROTECTED]/1
Thanks for the reply Bret.
I have tested this parsing issue ever way possible, equal variables unequal
variables, arithmetic, ie $[1 + 1], etc., etc., and my conclusion is that
Solaris does not parse the $[expr1 operator expr2] function properly. It
always produces a value of 0.
I installed 1.2b
I'm getting:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type:
IAX Subclass: REJECT
Timestamp: 9ms SCall: 8 DCall: 2
[66.98.180.77:4569]
CAUSE : No authority found
Under the DIDx number I'm putting IAX:
[EMAIL PROTECTED]/1567252(IAX)
where N is the rest of
I apologise if this is considered commercial, but I wanted people to know.
I've been fighting up with Asterisk and the TDM cards to hang-up when a
calling party disconnects, and I know I'm not alone in this fight!
My telco doesn't have polarity reversal for Disconnect Supervision, instead,
I get
Waldo Rubinstein ha scritto:
You mean to say that it will ONLY log if I have an h extension or if
I don't? Shouldn't it be logged no matter what?
No, of course it logs no matter whats, I was meaning that if you have
exten => h,1,...
exten => h,2,
ecc ...
don't expect the h extension to
Gurminder Arora wrote:
Hi raj,
Perhaps both of us are going through same tunnel...
Along with all the Zap users in India :) Occasionally there will be a
post in the list about Zap support for India, but even now there is no
CallerID or Call Progress monitoring for India. I know a bit of co
Just compiled from cvs head and enabled Real-time config
However, I now seem to have an odd problem, which I think
I’ve tracked to a patch in CVS head that adds domain support to SIP (http://bugs.digium.com/view.php?id=4466).
The problem is as following: -
Every time a sip peer/f
Thanks.I understand your POV. However, in addition to usage-based billing (which is what you refer to), I need to bill for the account. So, if the user placed two simultaneous calls with the same account, that may be fine because it could have been the call-waiting feature. However, if the user pla
You mean to say that it will ONLY log if I have an h extension or if
I don't? Shouldn't it be logged no matter what?
- Waldo
On Oct 11, 2005, at 5:31 AM, Simone Cittadini wrote:
Dinesh Nair ha scritto:
On 10/10/05 22:30 Waldo Rubinstein said the following:
1) When are asterisk CDR logs
Hi All,
- I have Digium cards and given that the archives point out the Digium
cards drop packets does anyone know what hardware would not do this?
(i.e. Allow me to send outbound faxes)
- If there is still an issue with the wctdm driver, does anyone know
which asterisk/spandsp combo would
Use virbiage IAX2 softphone.https://www.virbiage.com/download.php
Since it uses IAX2 it is very NAT friendly.
Thanks,
Steve
> Hi all I'm looking for a solution to this problem.
>
> *boxinternet---nat---softphone
>
> We have potential customers who will be trav
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