Hi There,
We install asterisk in RH9. We've T1 line and four TE410P card. We wanted to
configure this card.
We used ztcfg to see the channel configuration. It shows like
Channels 0. How to configure the channels and the four TE410P cards.
Regards
SipMonsters.
Hi,
-Original Message-
exten = 0,11,System(/scripts/sendSMS.pl -r 17083519199 -p
TMOBILE -s operator -m ${CALLERID}) exten = 0,12,Hangup
and this is the error:
-- Executing System(SIP/192.168.0.3-0891abc8,
/scripts/sendSMS.pl -r
17083519199 -p TMOBILE -s operator -m Bartosz
I just started to play with Asterisk today and while I'm
writing some IVR-like functionality in extensions.conf I
would like to take a decision based on whether playing a file
succeeds:
Use AGI() to either check for the file presence, or to determine
the rest of the dialplan logic
Try to comment out in your sip.conf
;qualify=yes
On Mon, 2004-08-09 at 06:52, Sathya Weerasooriya wrote:
Just wondering whether we have a resolution to iconnect incoming
problem, which started few days ago.
Cheers
SW
--
Best regards
Vlad
___
Hi Kevin,
no you didn't miss the reply and I've not resolved it yet.
Have you got similar problems?
Pamela
Kevin Fjelsted wrote:
Pamela,
Did you resolve the problems you described?
I didn't see a reply on the list but I may have missed it.
-Kevin
-Original Message-
From: Pamela Weis
--- Dennis Nacino [EMAIL PROTECTED] wrote:
Hi,
I just made it worked. The problem is brought about by the placing of odbc.ini and
odbcinst.ini on
/usr/local/etc. Anyway, I noticed with res_config_odbc in used, the * will only parse
the first
#include file it encounters. To illustrate;
my
Hello,
we have set up e164.lu as a test zone, as the
delegation for 2.5.3.e164.arpa hasn't been
completed yet. For all those who want to call the
numbers currently availble directly via SIP,
please use the zone name in your enum.conf.
If you decide to use the zone, please tell me at
[EMAIL
hi
we're a relatively new norwegian company terminating in norway. does
anyone know companies that terminate traffic around the globe? we've
got decent prices for .eu and .us, but we need cheaper solutions for
asia, middle east and africa.
regards
roy
Hello,
I am New user on Asterisk.. I have some problems;;
When I calledto another user from my user on soft phone, the call is correctly going, but when the other man receives the call and say "Hello", i can hear only first word and after that voice is not coming though call is going on.
Also
Hi,
It may be the problem of the CODECS that you are using in your configuration. Verify
your codecs.
On Mon, 09 Aug 2004 Nilesh sonavani wrote :
Hello,
I am New user on Asterisk.. I have some problems;;
When I called to another user from my user on soft phone, the call is correctly
Robinson Tim-W10277:
We are using the HFC card in point-to-point mode with DDI.
I am using bri-stuff-0.0.2 as well.
So, reading between the lines
To enable DDI/DID/DOD, the card must be run under the ZAPHFC channel driver
and therefore must be a HFC card? Can somebody confirm this?
I'm building a phone-in demo system to use for introducing Asterisk to
prospective clients.
One of the things I'm wary of is their likely preconceptions that VoIP
systems will have poor audio quality.
As a result, I'd like to ensure that the voice prompts I'm using have the
best possible audio
On Mon, 2004-08-09 at 01:36, Snak Newyork wrote:
Hi There,
We install asterisk in RH9. We've T1 line and four TE410P card. We wanted to
configure this card.
We used ztcfg to see the channel configuration. It shows like
Channels 0. How to configure the channels and the four TE410P
What I understood from earlier discussion is that the AVM cards do not
support ptp mode, only in the more expensive models. (Or was that Eicon,
but those are all expensive... mmmh) ;)
Either way, zaphfc/qozap seems to be the better choice for any application.
Nick Barnes wrote:
Robinson
Hi,
for incomming calls, i have set an gatekkeper in h323.conf.
outgoing calls wich are no sip endpoints should be sent to a h323
gateway, not the gatekeeper. If i disable the Gatekeeper, all non sip
calls are routed to the Gateway. If i enable the Gatekeeper, the calls
are send to the
Morning all
This is probably so simple but.
I have a need to set 3 values within each context ( extensions.conf ) .
It would appear that this can normally only be done when an exten is called
using SetVar / Global.
Is this right ? Can i set these values for use at any time ?
Why do i need to
Hello Again,
As you said It may be the problem with CODEC which i configured in my SIP.CONF.
I used followoing code for CODEC in SIPCONF file :
disallow=all ; Disallow all codecsallow=gsm;allow=g723.1;allow=ulaw ; Allow codecs in order of preference;allow=alaw;allow=gsm;allow=ilbc
;allow=ilbc
To enable DDI/DID/DOD, the card must be run under the ZAPHFC channel
driver and therefore must be a HFC card? Can somebody confirm this?
Basically yes, but ...
The reason I ask is that I installed a BRI system (Single Fritz! AVM
card using chan_CAPI) last week which refused to work
... if
Now the incoming from iConnect is working. The problem
was iConnect is not taking the Contact: header at
the time of registration. Though Asterisk is sending
the Contact: exten@x.x.x.x header the iConnect is
sending the call on phone #@x.x.x.x so all we need
to do is to remove the exten in the
Holger Schurig:
Basically yes, but ...
Many thanks for your help - I'll stop playing with the AVM cards now!
HFC cards are cheap as well.
Check the voip-info.org wiki, as usual :-)
Indeed. Had a look there and found a few cards, but what I really was after
was recommendations for a good
In article [EMAIL PROTECTED],
Steven Critchfield [EMAIL PROTECTED] wrote:
On Mon, 2004-08-09 at 06:07, David Gurr wrote:
I'm building a phone-in demo system to use for introducing Asterisk to
prospective clients.
One of the things I'm wary of is their likely preconceptions that VoIP
I'm interested to hear how folks are handling NAT SIP routing issues in the
wild for commercial use.
Are you using a commerical SIP-aware NAT router solution? If so, what?
Are you using a software SIP-proxy like SER or siproxd? If so, which?
Do you set everything to canreinvite=no in sip.conf?
On Monday 09 August 2004 14:06, Nick Barnes wrote:
Holger Schurig:
Basically yes, but ...
Many thanks for your help - I'll stop playing with the AVM cards now!
but chan_capi is in sync with * cvs, hfc-s support (bri_stuff) no
--
Maurizio Marini
with a h323 client over my gatekepper a call comes over asrerisk to my
sip endpoint:
== Spawn extension (sip-phones, 01634255122, 1) exited non-zero on
'SIP/0699073201-528d'
-- Executing Dial(H323/ip$10.0.0.124:49638/18690,
SIP/0699073201) in new stack
-- Called 0699073201
--
On Sun, 8 Aug 2004 15:04:41 -0400, Steve Szmidt [EMAIL PROTECTED] wrote:
Here's a version I modified which grabs either a development or stable
verision, and does a backup before updating from CVS. It also asks for
addon's and cc.
Leif Madsen did the original development and Mark released
Hi list,
when I put a call in parking and take it back, I'm not able to put it
again in parking. Context is empty and I receive message that extension
7 (or 70 if I'm quick) is not existing. Is this a bug or misconfiguration?
Cheers
--
Daniel
___
Hi all,
Hello !!
I saw in FWD site a phone on the web.. (click 612 link)
http://www.freeworlddialup.com/advanced/beta_programs
I´d like to have this application in my intranet.. click on my name, than
calls my number..
I´d also like to see that phone on the web... as an option
How can I do that ?
Is it
dirk los [EMAIL PROTECTED] wrote:
I try to make an asterisk system and downloaded and unzipped the file
asterisk-1.0-RC1.tar.gz. When I do the first make I got the following
messages: .
checking for tgetent in ltermcap...no
checking for tgetent in ltinfo...no
checking for
Hello!
Am Sonntag, 8. August 2004 20:12 schrieb Steven Critchfield:
But * shouldn't crash with a core dump if mpg123 crashes anyway. mpg123
dumps the decoded stream to stdout (-s) and it might have some problems
with id3 tags.
So could it have just been that your music on hold pointed to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 09 August 2004 09:04 am, Leif Madsen wrote:
That script is pretty old now, so I'm glad someone is going through
and updating it. I am a bit confused by the statement ...and Mark
released it as I don't know where it got released. Is it
hi agian,
i am pondering why no one ist answering to thiis problem. i found 3
list-useres who have all the same problems, but ei can not find any
solution for that.
wenn ich do a call to 1234 with a h323 softohone to a sip endpoint all
works fine. If i make a call from PSTN to the same sip
I still have the problem, but have done a little further
isolation.
First, if there is no outbound iconnect section in
sip.conf, my incoming calls work fine (as long as my "register" statements exist
in the top section).
But, when I add an outbound section, using either 'peer' or
I TRIED , SAME PROBLEM.
The value doesnot have any characters, but script fails.
Have no idea why.
Bart
Hi,
-Original Message-
exten = 0,11,System(/scripts/sendSMS.pl -r 17083519199 -p
TMOBILE -s operator -m ${CALLERID}) exten = 0,12,Hangup
and this is the error:
--
I TRIED , SAME PROBLEM.
The value doesnot have any characters, but script fails.
Have no idea why.
Bart
Hi,
-Original Message-
exten = 0,11,System(/scripts/sendSMS.pl -r 17083519199 -p
TMOBILE -s operator -m ${CALLERID}) exten = 0,12,Hangup
and this is the error:
--
Who are the US wholesalers selling the uniden phones?
Thanks,
Gary
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I'm trying to use the ChangeMonitor command on the asterisk manager API, but I
can't find the syntax anywhere. Asterisk only tells me:
Action: ChangeMonitor
But I don't know the parameters. Can anybody help me?
___
Asterisk-Users mailing list
[EMAIL
On Mon, 2004-08-09 at 08:34, Kevin Walsh wrote:
dirk los [EMAIL PROTECTED] wrote:
I try to make an asterisk system and downloaded and unzipped the file
asterisk-1.0-RC1.tar.gz. When I do the first make I got the following
messages: .
checking for tgetent in ltermcap...no
On Mon, 2004-08-09 at 11:24, [EMAIL PROTECTED] wrote:
I'm trying to use the ChangeMonitor command on the asterisk manager API, but I
can't find the syntax anywhere. Asterisk only tells me:
Action: ChangeMonitor
But I don't know the parameters. Can anybody help me?
It takes two
Hi,
I have read quitea bit of the available resources and have this idea of
asterisk. Would someone kindly answer these briefly
1-) Asterisk does not need a sound card...but if i am to record voice into
an extension or dial from CLI ( basically use asterisk itself as a softphone
) then i need a
niko singh wrote:
Hi,
I have read quitea bit of the available resources and have this idea
of asterisk. Would someone kindly answer these briefly
1-) Asterisk does not need a sound card...but if i am to record voice
into an extension or dial from CLI ( basically use asterisk itself as
a
On Mon, 2004-08-09 at 12:18, niko singh wrote:
Hi,
I have read quitea bit of the available resources and have this idea of
asterisk. Would someone kindly answer these briefly
1-) Asterisk does not need a sound card...but if i am to record voice into
an extension or dial from CLI ( basically
Here is a sample call file that I am using:
MaxRetries: 2
extension: 9997
Channel: IAX2/USERID:[EMAIL PROTECTED]/14037422000
CallerID: LAKEVIEW 4037422000
Anyway, this works fine. The problem is that specifying the channel
this way does not handle problems very well. If hagenhomes is down,
the
try help application changemonitor in the Asterisk CLI
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I got it.
My fault. I should read the instructions better.
Yum installed 123mpg which does not work with asterisk.
I reinstalled the old version and it is ok.
Bart,
Yesterday, I did update my server with some packages.
After that music on hold is playing very slowly.
Rest works fine.
This
I'm interested to hear how folks are handling NAT SIP routing issues in the
wild for commercial use.
Are you using a commerical SIP-aware NAT router solution? If so, what?
Are you using a software SIP-proxy like SER or siproxd? If so, which?
Do you set everything to canreinvite=no in sip.conf?
Gary Carr wrote:
James H. Thompson wrote:Who are the US wholesalers selling the uniden phones?
www.thevoipconnection.com
But unfortunately they are on backorder
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can you use .wav files or does it have to be gsm?
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 09, 2004 4:23 AM
Subject: Re: [Asterisk-Users] Sound file quality
On Mon, 2004-08-09 at 06:07, David Gurr wrote:
I'm building a
Just write a CGI script that places a file in in the outgoing calls
directory. /var/spool/asterisk/outgoing, I believe. This will
accomplish what you're wanting.
-g
On Mon, 2004-08-09 at 11:45, Andrew Thompson wrote:
Andrei Goncalves wrote:
Hello !!
I saw in FWD site a phone on the
Glen Hinkle wrote:
Just write a CGI script that places a file in in the outgoing calls
directory. /var/spool/asterisk/outgoing, I believe. This will
accomplish what you're wanting.
Did you even click the link?
I saw in FWD site a phone on the web.. (click 612 link)
I'm in the process of doing the same thing. My approach is to declare
asterisk as h323 gateway for the Cisco Call Manager, then define a route
pattern to call asterisk. The strange thing that i'm dealing with now
is, that the inbound RTP stream is going from the phone directly to
asterisk and
On Mon, 2004-08-09 at 12:57, Holger Schurig wrote:
try help application changemonitor in the Asterisk CLI
I'm sure you meant show application changemonitor. That will show the
dialplan application ChangeMonitor but not the Manager API ChangeMonitor
command. The show manager command ChangeMonitor
On Mon, 2004-08-09 at 13:26, Andrew Thompson wrote:
Glen Hinkle wrote:
Just write a CGI script that places a file in in the outgoing calls
directory. /var/spool/asterisk/outgoing, I believe. This will
accomplish what you're wanting.
Did you even click the link?
I saw in FWD site
Hello.
I have a very strange H323 problem. This is the situation: I have a
Cisco 7960 phone (with IP address 10.1.1.21) connected to a Cisco
CallManager (with IP address 10.1.1.10) and an Asterisk with IP address
10.1.1.22. I have managed to make the CallManager to call to asterisk
using a
Raj, yes your post helped me.
Just to complete the whole thing and clarify the problem that was
posted by Greg Blakely;
First, if there is no outbound iconnect section in sip.conf, my incoming
calls work fine (as long as my register
statements exist in the top section).
But, when I add an
Steven Critchfield wrote:
.
As a seperate option, the CGI solution above kind of gets a similar
functionality. No it doesn't use the web browser, but it would allow you
to collect a phone number and issue a call out to the person requesting
the call. You then could select when and how to connect.
Followed the instructions on voip-info.org regardinging fedora FC2,
making Zaptel seems to work fine, however when I modprobe I get this. It
looks like a version mismatch somehow. Ideas? If this ooc, sorry first
post here :-)
modprobe tor2
WARNING: Error inserting zaptel
Is it possible to limit which Zap channels answer the phone? If I have four
numbers but only want asterisk to answer the 1st channel, and allow all four
channels out bound in a hunt group. I think this is called a trunk group,
does it support this? Any information would be helpful.
[EMAIL
And here I was trying to figure out how to kill the blinking display :-)
OK - dumb newbie award hereby rewarded to me. Thanks. And I had already
checked the wiki and done what you suggested in sip.conf - so my
stupidity wasn't total :-)
Stupidity may be a bit strong in any case. The real
Steven Critchfield wrote:
On Mon, 2004-08-09 at 13:26, Andrew Thompson wrote:
Glen Hinkle wrote:
Just write a CGI script that places a file in in the outgoing calls
directory. /var/spool/asterisk/outgoing, I believe. This will
accomplish what you're wanting.
snip
As a seperate option,
Hello Matt,
I had the same problem ...
Changing the linux-2.6 symlink in /usr/src to /lib/modules/2.6.5-1.358
(with kernel-2.6.5-1.358) and building it again with make clean; make
linux26 made it work (so the symlink is /usr/src/linux-2.6 -
/lib/modules/2.6.5-1.35).
Cheers,
Oliver
Matt Schulte
In article [EMAIL PROTECTED],
hank [EMAIL PROTECTED] wrote:
can you use .wav files or does it have to be gsm?
You can use .wav files. They should be PCM format, 8000Hz sampling,
16 bit mono. Windows Sound Recorder can produce them, as can sox.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL
Thank you very much Jeremy. This works perfectly. I have been
struggling for a long time to get this to work but it now works
perfectly. I have added a comment to the wiki:
http://www.voip-info.org/wiki-Asterisk+auto-dial+out#comments
Thanks Again,
Darren Wiebe
[EMAIL PROTECTED]
Jeremy Hall
hank wrote on 8/9/04, 12:10 PM:
can you use .wav files or does it have to be gsm?
See the wiki
http://www.voip-info.org/tiki-index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk
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Actually, they are in stock now. At least they were able to fill waiting
orders. Mine came in today.
-Nate
-Original Message-
From: Paul Zimm [mailto:[EMAIL PROTECTED]
Sent: Monday, August 09, 2004 12:07 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] uniden phones
Gary Carr
Hi all,
Is it just me and not reading the docs right, or has anybody else had
problems with the AbsoluteTimeout application and the 'T' extension when
used inside a macro?
[macro-attended]
; ARG1 is the device to dial out on, SIP or Zap, or whatever
; ARG2 is the extension to dial using
I did that, now I get this error on compile:
make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/lib/modules/2.6.5-1.358'
make[1]: *** No rule to make target `modules'. Stop.
make[1]: Leaving directory `/lib/modules/2.6.5-1.358'
make: *** [linux26] Error 2
Andrei,
It is an activeX control and it no longer needs
to have the security level changed.
(The earlier version was an unsigned applet that required the security
level change until I bought the code signing certificate early this year.)
If you want to have a soft phone running in a web
Can anybody suggest how to setup company having 3 bri connections to local
telco.
As far as I understood iax is supper good for interconeting branch
offices. quadBRI is the best solution if you have more than 1 ISDN
channel. But every company still needs a fax. More or less hylafax is
suitable for
I agree with you there, I wouldn't feel too stupid, the same thing happened
to me when I purchased my BT101... The picture on the website and on the box
shows the message button lit up in red, I naturally assumed that when MWI
was triggered, that would happen... but I quickly realized that it was
No, i made a mistake ... the symlink is actually linux-2.6 -
/lib/modules/2.6.5-1.358/build
I forgot the build - I am very sorry about that.
Matt Schulte wrote:
I did that, now I get this error on compile:
make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory
This evening I tried to install asterisk RC1 .rpm on a Fedora box... but how
can I re-add openh323 support? or does it contain an alternate h323 support?
thanks in advance
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use the fourth bri span with a BRI TA with 2 FXS..
3Com/USR has a working model, but there are plenty more.
[EMAIL PROTECTED] wrote:
Can anybody suggest how to setup company having 3 bri connections to local
telco.
As far as I understood iax is supper good for interconeting branch
offices. quadBRI
Should I be concerned about this? It seems to only happen
when my MoH switches songs. The songs sound as good as an 8k/s song would.
Travis Conway
EFS, Inc.
Information Technology
Desk: (334) 215-6551
Mobile: (334)
391-4450
mailto:[EMAIL PROTECTED]
is there something that needs to be set up to make the 'called' and
'callers' buttons work on this phone?
all i get is the backlight to switch on and off.
Jason Kawakami
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I am trying to get one of the function keys on the Snom 200 working as an
intercom. However, I can't get the other Snom 200 phone to auto-answer. I
found some posts in the archives from Christian that talk about intercom=true
and also the Call-Info header. However, I can't get either one to
I have searched all over the web and have not really found anything
related to this error The only thing I found is related to a
system stops responding on DTMF, which does not happen here... THe
following is the output from the CLI:
*CLI 2004-08-09 17:36:29 DEBUG[229390]:
Thomas Kuepper wrote:
Hi,
for incomming calls, i have set an gatekkeper in h323.conf.
outgoing calls wich are no sip endpoints should be sent to a h323
gateway, not the gatekeeper. If i disable the Gatekeeper, all non sip
calls are routed to the Gateway. If i enable the Gatekeeper, the calls
I am having problems getting the latest CVS right now. A cvs checkout asterisk -t
gets to this part and sits forever:
S- server_register(fpm-world-mix.mp3, 1.1, , , , , )
S- Register(fpm-world-mix.mp3, 1.1, , , )
Anyone know how I can just skip the file?
Travis Conway
EFS, Inc.
Information
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 09 August 2004 05:58 pm, rayers.list wrote:
I am trying to get one of the function keys on the Snom 200 working as an
intercom. However, I can't get the other Snom 200 phone to auto-answer. I
found some posts in the archives from
Or you can install the kernel 2.6.7 and all those little worries
disappeared. I don't know what they did in FC2 to get it so wrong with
their kernel...
Jean-Yves
On 10/08/2004, at 5:37 AM, Oliver wrote:
I had the same problem ...
Changing the linux-2.6 symlink in /usr/src to
On 9 Aug 2004 at 12:35, Joseph wrote:
respectfully, Joseph ===
-= Psalms 9:17 =
Woah!
The wicked shall be turned into hell, and all the nations that forget
God.
Bit intense for an asterisk mailing list!
:-)
Matt Riddell
I just ended up reverting to 2.4.22-1.2188.nptl. Nothing really all that
interesting in 2.6.x for a production server yet, for me anyway.
-Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Yves
Avenard
Sent: Monday, August 09, 2004 8:30 PM
To:
David == David Gurr [EMAIL PROTECTED] writes:
David As a result, I'd like to ensure that the voice prompts I'm
David using have the best possible audio quality.
David My callers will be coming in over PSTN to a VoIP gateway and
David then to me by uLaw/aLaw ...
The optimal quality in the case
Yes, I read that. It is not in there. It does mention setting the
Auto-Answer for the phone. However, I want an intercom, I don't want a door
phone. The Auto-Answer feature just sets it so it answers all calls
automatically.
Ryan
-Original Message-
From: [EMAIL PROTECTED]
Hello-
I am new to the list, so forgive me if this has been answered, but I
haven't seen it on google yet.
What I want to know is, is there any way to send a hook-flash signal from a
cell phone (and then have Asterisk pass it up to the PSTN?)
I suspect we need an example.
I have an
rayers.list wrote:
I am trying to get one of the function keys on the Snom 200 working as an
intercom. However, I can't get the other Snom 200 phone to auto-answer. I
found some posts in the archives from Christian that talk about intercom=true
and also the Call-Info header. However, I can't
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 09 August 2004 11:37 pm, Michael Welter wrote:
In the same vein, I would like the proper button to light when I call an
extension. I have five extensions configured on the SNOM, 201-205.
When I dial 203, as an example, the top button
On Mon, 9 Aug 2004 15:25:02 +0200, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi all,
I have 'Voipac NetPhone 210' phone apparatus with H.323 support
Is there any way to connect it to asterisk?
What exactly I need to do.
Thank you very much.
hi,
Just google for oh323 and the first
Hi,
I've looked through the list archives, bug tracker and cvs changelogs
and can't see anything that refers to the particular problem i've seen
recently.
I'm running CVS-D2004.06.09.14.00.00-06/24/04-00:43:55 which I
realise is not exactly recent, but I wanted to find out more
before I updated.
Hi, Ed Guy,
(B
(BDo you find interesting issue when getting CDR?
(BIt has two billing leg on "click to call". One is origination leg and the
(Bother is termination leg. After call drop, only termination leg CDR is
(Blogged.
(B
(BRegards,
(Byang
(B
(B
(B-Original Message-
Greg Hill wrote:
On Sun, 8 Aug 2004, Kevin Johnson wrote:
I'm having a problem with extensions.
Any extension longer than 6 characters gets truncated to 6 characters.
For example,
exten = _7XX,3,NoOp(call for${EXTEN})
results in
call for 712345
when given
7123456
that's ${EXTEN}, not
Hey there,
I don't know who else has suffered broadvoices terrible service, but I
am about to my end with them. The lack of a LBR codec, the outages,
the changing of servers without notifying subscribers haspushed me to
my end. Now most incoming calls are abbruptly cut off within a minute
of the
gafachi
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