=${SIP_HEADER(To)})
Set(DailedNumber=${CUT(ToHeaderVal,:,2)})
Set(DailedNumber=${CUT(DailedNumber,@,1)})
That should give you the dialed number in Variable "DialedNumber".
Greetings
Max
Am 22.12.2017 um 14:54 schrieb Benoit Panizzon:
> Dear List
>
> It looks like the c
ith "stuck" DNS.
But that was ages ago... The last time on my old "Horstbox" with Asterisk 1.2
and bristuff on Linux 2.4 :-/
Have you rebooted the whole WRT device or just restarted the Asterisk service
to resolve your problem?
Maybe it's less an Asterisk issue but one with D
Hello,
I'm also a customer of the DTAG.
Yesterday, the messed a bit with their DNS entries...
If you are NOT using their DNS resolvers you got a "wrong" IP address back that
was not working.
Besides that, you should disable SRV lookups for their SIP peers. Since
Asterisk's chan_sip.c does not
ions - you should
rely on what your telephone number provider tells you to do ;-)
Greetings
Max
Am 28.03.2017 um 15:24 schrieb Olivier:
> Hello,
>
> In France, years ago, there was some discussions about a new regulation
> forcing some providers to not charge anything
arrier should fix this problem on his site.
If he wants to enforce rate-limiting to INVITEs he should do it right by
honouring the Call-IDs and sequence numbers.
If you like you can anyway send me your trace off-list, maybe there's something
other weird going on.
Greetings
Max
Am 22.02.2017 um
ll try again with the bundled version.
If the bugs persist, I'll file some bugs ;-)
Max
signature.asc
Description: OpenPGP digital signature
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check
SIP
in other software), but after just five minutes of testing
I found several bugs regarding PJSIP preventing me to use it in a production
enviroment :-(
I'm going to file these bugs at the moment...
Max
signature.asc
Description: OpenPGP digital signat
for testing purposes ;-)
Max
Am 15.02.2017 um 19:46 schrieb Motty Cruz:
> Hello, I have a user that prefers Soft SIP phone install on his laptop, for
> security reasons I have enable TLS on our Asterisk server to support TLS
> authentication, It works well with hard phones. Has anybody in this
e in sip.conf resolves to a CNAME and I
> change the CNAME in my DNS?
Not as far as I know.
If you enabled SRV lookups for Asterisk, you may also want to check possibly
existing SRV records for your host since Asterisk then l
auth_options_requests=yes
---
Greetings
Max
Am 08.01.2017 um 19:47 schrieb Luca Bertoncello:
> Luca Bertoncello <lucab...@lucabert.de> schrieb:
>
> Hi again!
>
>> The problem: after 15 minutes will the call dropped, but only if the call is
>> to another nation
Hi,
That's right - you just need to define a peer with a static IP address and
"type=peer" to assign incoming calls to a peer name and apply
the corresponding configuration (e.g. codecs).
To make your configuration less redundant you can use templates in your peer
definition
(at least for
uot;externip" and "localnet" is all you need to
help Asterisk setting the SDP address correctly.
Also, enabling ICE support can help you getting the correct IP address if the
remote peer supports it.
Greetings
Max
Am 07.12.2016 um 00:02 schrieb Harel:
> Hello List,
&g
I made my best results with all devices, sadly it's not very
common used.
Max
Am 23.11.2016 um 20:02 schrieb D'Arcy Cain:
> On 2016-11-22 07:49 PM, Pete Mundy wrote:
>>
>> One direction that may be worth exploring further is his ATA's config (or
>> perhaps swapping it
sub with only the (unique) reference name of the
variables you wish to pass and then call it from your Gosub.
-> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_SHARED
Greetings,
Max
Am 23.11.2016 um 13:06 schrieb Jonathan H:
> Related to
> http://lists.digium.com/piperm
it makes things better.
If it does, you can prevent res_timing_dahdi from being loaded in your
modules.conf.
Max
signature.asc
Description: OpenPGP digital signature
--
_
-- Bandwidth and Colocation Provided by http://www.api-digi
idged between two ends
of a call.
Asterisk normally synchronises the RTP clocking to one end of the call. But if
this RTP source is not realiable (jitter, packet loss, silence suppression...)
you
can end up having audio problems.
Max
e not disabled re-invites, the RTP address may
change while the call is running.
Max
signature.asc
Description: OpenPGP digital signature
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out t
oice
stream.
So - this is definetely worth to investigate and to get your ITSP have a look
at it. There are many ways to stop other customers from doing this (maybe this
happens accidently).
If you have further questions you might contact me off-list - since this is
something that does not reall
d eleminate all characters you're not expecting.
Greetings
Max
signature.asc
Description: OpenPGP digital signature
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community
Hi,
some phones can add a pause when dialing, sometimes by holding the * or # key a
few seconds after the first digit.
If it works, the phone normally adds a "W" or ";" to the dial string.
So you would program the speed dial key with <*2[hold * or #]101>.
Am 01.10.2016 um 20:22 schrieb Tech
be external,
so you may see a second call starting at the time where the client left the
bridge.
Max
signature.asc
Description: OpenPGP digital signature
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Maybe the client just put the call on hold.
So the call technically has not ended AND the client does not need to
send or handle any RTP data.
Is there any mention of "music on hold" for this channel?
Greetings
Max
- Nachricht von Leandro Dardini <ldard...@gmail.com>
matched "exten" pattern is
stored (i.e. the "_+49555.")?
I just need something unique to find the dialed extension in the table...
Thanks!
Greetings,
Max
pgpYSs8pSplWt.pgp
Description: Digitale PGP-Signatur
--
__
required, but missing table field) causes
this issue.
But even if not, you are getting rid of the warning messages ;-)
Greetings
Max
signature.asc
Description: OpenPGP digital signature
--
_
-- Bandwidth and Colocation Provided by
ed call
diversion) is normally stored in REDIRECTING(from-num).
[1] https://wiki.asterisk.org/wiki/display/AST/Function_REDIRECTING
Greetings
Max
signature.asc
Description: OpenPGP digital signature
--
_
-- Bandwidth and Coloca
On 11.06.2013, at 0:24, Sean Darcy wrote:
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no
success:
Silk is enabled only after asterisk restart.
for silk work need codecs.conf with silk configuration
res_format_attr_silk.so - loaded
codec_silk.so - loaded
please
TE420B.
Can anybody provide me required ss7.conf file and also provide dahdi
configuration which is needed for this device.
Thanks you so much in advance!!
Thanks,
Max Alex
--
_
-- Bandwidth and Colocation Provided by http://www.api
37pgn.
http://darkskiesblog.com/wp-content/uploads/img/vosc.html
22gv7con3pr fjfvojvm e1cugvfj, tuzj2tcz2 4zssju6k5bfj. jdc52cc
twdoh35sn4s sb2yj.
--
Thanks,
Max Alex
Voip Developer
--
_
-- Bandwidth and Colocation Provided
but i am not able to get their
audio,
I have disabled firewall, selinux is also off.
If I am applying this line to analog phone then also it is working fine,
But when it is added on digium card then this issue happens,
can anybody help me for this issue?
Thanks,
Max Alex
Voip Developer
Hi,
Thanks for this information, but it is not working for both the issues,
I have tried with the configuration with cidsignalling, cidstart etc..
Can any one provide more help for this.
Thanks,
Max Alex
Voip Developer
On Mon, Aug 9, 2010 at 5:31 PM, asteriskguru asteriskguru
beaasteriskg
callerid=asreceived
signalling = fxs_ks
channel = 3
context=from-zaptel
group=0
echocancel=yes
callerid=asreceived
signalling = fxs_ks
channel = 4
---
Please hemp me for this issues.
Thanks,
Max Alex
Voip Developer
2001 put on hold to 1001.
Please let me suggestions on this.
Thanks,
Max Alex
Voip Developer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
idea what this is
an how to configure it to a restricted range of IP addresses?
Nicholas Blasgen
Partner / Network Operations
Refractive Dialer LLC
(724) 252-7436
On Sun, Jan 3, 2010 at 8:29 PM, Max McGraw wrote:
Nicholas,
you haven't specified which version, which does make
a lot
hello,
we have been using a couple of US based
VoIP providers for outbound calls completed
within the US, without any issues.
We recently started making calls to Canada
and have received a few complaints about
the call quality.
Questions :
- Could this be because of the number of
On Tue, Jan 5, 2010, UIT DEV wrote:
Steve-
Got an iPhone [...]
As I got to reading I began to see things like provider, as you've
said here, and unfortunately if that is the only way then I will have
to stop here as I do not have funds to further this little experiment.
I guess I
at 11:25 AM, Max McGraw max.mcg...@gmail.com wrote:
hello,
we have been using a couple of US based
VoIP providers for outbound calls completed
within the US, without any issues.
We recently started making calls to Canada
and have received a few complaints about
the call quality
income - its not an option.
Hopefully you'll not encounter sucky times, else you'd know.. That
couple of bucks a month will never just be a couple of bucks.. :-)
On Tue, Jan 5, 2010 at 4:47 PM, Max McGraw max.mcg...@gmail.com wrote:
On Tue, Jan 5, 2010, UIT DEV wrote:
Steve-
Got
Nicholas,
you haven't specified which version, which does make
a lot of difference.
1.6.x can easily traverse NAT. If you are only making
outbound calls, you shouldn't need to forward 5060.
Unless you have a special NAT that is blocking
outbound connections, the SIP.conf settings
for this?
Thanks in advance!!!
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
Ok, so I've made progress on 2BCT (2 B-Channel Transfer). I'm assuming that
the debug info below shows that XO doesn't have 2BCT enabled on my line, but
if anybody can confirm that'll let me be way more indignant. J
-- Native bridging DAHDI/1-1 and DAHDI/3-1
Protocol Discriminator:
I'm trying to get blind transfer from an incoming DAHDI line to an
external number to work on an * 1.6 install using a T1 from XO. The
documentation is very distributed and incomplete, so while it's not
working, it's definitely more likely my error somehow. Couple questions if
anybody is out
(${DIALSTATUS})
Thanks,
Max Alex
Voip Developer
On Wed, Apr 8, 2009 at 9:47 PM, Klaus Darilion klaus.mailingli...@pernau.at
wrote:
Haven't you read my email?
1. Wrong list
2. Missing log entries (set debug 4, set verbose 4)
klaus
Max Alex schrieb:
Hi All,
Thanks for your reply
%3a7...@192.168.1.25
Call-ID: 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
CSeq: 34526 REFER
User-Agent: Grandstream BT200 1.1.6.46
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
-
--- (14 headers 0 lines) ---
Call
Hi
I have used the transfer operation this way.
When i got a call on grandstream phone, i will receive it
and press transfer button and enter transfer number and press send button.
My call is disconnected but no call transfer from asterisk.
Please advice me!!
Thanks,
Max Alex
Voip Developer
transfer button of the grandstream phone.
Can anybody provide help for this issue?
Thanks in advance!!
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE
the cause of crash?
Please checkout following link, I have uploaded coredump backtraces there.
http://pastebin.com/m5480bcb8
Please provide me help regarding this.
Thanks in advance.
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided
have put the incoming call on hold and when we try to resume it back,
the call is hangup, and not able to connect the hold channel.
Can anyone provide help!!
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api
in Advance!!
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
a problem with asterisk 1.6 deadagi application, when the call is
hangup at that time the script is exited and no duration and status will be
counted, So please provide help regarding this deadagi application in
asterisk 1.6 branch,
Please help me regarding this!!
Thanks in Advance!!!
Thanks,
Max Alex
Voip
.
Please provide information!!!
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
provider as they have an issue more likely
though is you have a bad dial command.
/telepathy
--
Kind Regards
Max Brooks - Developer
Legatio Technologies Limited
Phone: 01793 520 506
www.legatio.com, www.ftax.co.uk
Legatio is part of the Callcredit Information Group: www.skipton.com
into the / partition and run passwd.
--
Kind Regards
Max Brooks - Developer
Legatio Technologies Limited
Phone: 01793 520 506
www.legatio.com, www.ftax.co.uk
Legatio is part of the Callcredit Information Group: www.skipton.com,
www.callcredit.co.uk, www.eurodirect.co.uk
Legatio Technologies
disappered and we must have to reload to load moh again.
Can any body please help me regarding MOH configuration!!
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
.
Thanks,
Max Alex
Voip Developer
On Mon, Jan 12, 2009 at 6:45 PM, Philipp Kempgen
philipp.kemp...@amooma.dewrote:
Philipp Kempgen schrieb:
Max Alex schrieb:
If i got the NOANSWER then the channel is not passing to next priority.
I need to pass that channel to the next priority
Hi All,
We have already use 'g' option in that, but it is not working in my case.
Thanks,
Max Alex
Voip Developer
On Sun, Jan 11, 2009 at 1:51 AM, Philipp Kempgen
philipp.kemp...@amooma.dewrote:
Max Alex schrieb:
If i got the NOANSWER then the channel is not passing to next priority.
I
)
exten=s,1,Goto(${MACRO_EXTEN}|1)
[macro-voicedid]
exten=s,1,NoOp(${ARG1})
Please provide me help regarding this!!!
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
itself or we have to setup some preventions for that.
Can anybody suggest me regarding this freeze cli issue!
Thanks in advance!!
Thanks,
Max Alex
Voip Developer
On Wed, Jan 7, 2009 at 7:07 PM, Grygoriy Dobrovolskyy
megaho...@gmail.comwrote:
2009/1/7 Max Alex max.aster...@gmail.com
Hi
Hi,
Thanks for your reply
Can you suggest me how can we avoid it by doing any configuration changes in
asterisk.
So the freeze issue may not be occurred again!
Please provide me some help!!!
Thanks in advance!
Thanks,
Max Alex
Voip Developer
On Wed, Jan 7, 2009 at 12:58 PM, Grey Man greymanv
.
And because of this my iaxmodems are also getting time out from asterisk.
Please provide some help regarding this freeze issue.
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
HI,
Thanks for your reply,
But we have not setup DNS servers in asterisk. Asterisk is not getting any
DNS requests.
Please provide help regarding this.
Thanks,
Max Alex
Voip Developer
On Tue, Jan 6, 2009 at 4:10 PM, Grey Man greymanv...@gmail.com wrote:
Make sure the DNS servers Asterisk
can not allow 2102 not to
forward on 2103.
and also i want to prevent the SIP/2.0 302 Moved Temporarily.
please advice me that how can we set the user for not to forward or transfer
on 2103.
i have tested with allowtransfer=no in sip.
Thanks in advance!
Thanks,
Max Alex
Voip Developer
services so our calls will not be disconnected and recieved by them.
Please provide some help for this.
Thanks,
Max Alex
Voip Developer
On Sun, Nov 30, 2008 at 1:07 AM, Philipp Kempgen
[EMAIL PROTECTED]wrote:
Max Alex schrieb:
Actully we are getting the anonymous callerid from
for this!
Thanks,
Max Alex
Voip Developer
On Fri, Nov 28, 2008 at 7:47 PM, Philipp Kempgen
[EMAIL PROTECTED]wrote:
Max Alex schrieb:
I have one issue regarding override callerid when i have anonymous call.
I have added PAI in sip header and also set sendrpid = yes in sip.conf
Hi All,
I want to prevent transfer on based of user,
means we can disable any user or peer to transfer calls in asterisk.
Can any one helps how can we prevent transfer feature.
I am using asterisk 1.4 branch.
Thanks,
Max Alex
Voip Developer
using asterisk 1.4 branch.
thanks in advance!!
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman
in eye beam.
and in asterisk i got Method is not implemented.
Can anybody helps me in this?
If any patches are there then please let me know.
Thanks in advance!!
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api
Hi All,
I am using asterisk 1.4.22 in my local system
I want to know how can we set ability to log and report RTP and jitter
statistics per call.
Is there any configuration in logger or configuration in rtp?
Please provide some guide lines for this.
Thanks in advance!
Thanks,
Max Alex
Voip
Hi All,
Thanks for reply
i have tried for this, it looks fine for me,
but is there any way to check rtp log while call is connected or any way to
enable it to write in log file.
Please give me some guide lines!
thanks in advance.
Thanks,
Max Alex
Voip Developer
On Sat, Nov 15, 2008 at 3:21 AM
Hi All,
I am checking srtp support in asterisk 1.6,
Let me know any patches available or changes needed for srtp support in
asterisk 1.6.
Thanks in advance!
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api
hello,
I am using sip, my default codec is set to gsm in sip.conf
Using call files, is there a way to send out a call using
ulaw while other channels are using gsm ?
tia.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
hi, using sip, my default codec is set to gsm in sip.conf
I occasionally want to send out a call using ulaw while other channels
are using gsm, how can I do this using call files ?
I couldn't find any codec parameter in the call file definition.
tia.
,Hangup()
exten = h,1, Noop (PC - Hangup)
; HERE I LIKE TO LOG if destination pressed ** or just hangup
exten = h,n,system(/ pc /bin/ log_call ${ DIALEDTIME }:::${ ANSWEREDTIME
}:::${ PC_STATUS }:::${ HANGUPCAUSE }:::${ DIALSTATUS } )
Regards Max
=z9hG4bK-23a4ba1;rport
From: Anonymous sip:[EMAIL PROTECTED];tag=89cc6491fcf8ae21o1
To: sip:[EMAIL PROTECTED]
Remote-Party-ID: sip:[EMAIL PROTECTED];screen=yes;privacy=full;party=calling
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Anonymous sip:[EMAIL PROTECTED]:5061
Expires
to functions and using that functions in dialplan.
but it is always gives me function is not registered.
can any body explain how to register custom functions in asterisk?
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http
Hi,
can you please confirm that DTMF is working properly or not?
Thanks,
Max Alex
Voip Developer
On Sat, Sep 27, 2008 at 12:24 AM, equis software [EMAIL PROTECTED]wrote:
Hi, when I make a call I need that the caller can** hang up by dialing ***(H
option in Dial command), the call
Hi Hiren,
Can you please confirm the php-gd is properly installed?
Thanks,
Max Alex
Voip Developer
On Tue, Sep 9, 2008 at 4:20 PM, Hiren Mistry
[EMAIL PROTECTED]wrote:
Dear All,
I have configured here Asterisk-stat (Call Detail Records)for
CDR ANALYSER. Here I am facing
and suddently asterisk crashes
and i can't get email notification for received faxes.
any one help me about the crashes of asterisk?
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008
Hi,
let me know that you have configured properly in res_pgsql.conf in asterisk
with proper, and it is connected properly to database with database details.
Thanks,
Max Alex
Voip Developer
On Fri, Aug 29, 2008 at 10:26 AM, Hiren Mistry
[EMAIL PROTECTED] wrote:
Hi ,
I have check zapte.conf
Hi Hiren,
Have you properly configured the zap channels in asterisk,
which device have you configured in asterisk with zaptel?
let me know the dial plan for ivr.
Thanks,
Max Alex
Voip Developer
On Thu, Aug 28, 2008 at 11:40 AM, Hiren Mistry
[EMAIL PROTECTED] wrote:
Hi, Everybody,
I am
for this?
--
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update
is also played, and dtmf is also set
properly.
But i am not getting why the incoming call is not transfer to any other
number?
Please help for this issue!
--
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by http://www.api
calls?
Regards,
Max
On Jan 11, 2008, at 14:41 , pgck nirukshitha wrote:
Hi All
I am getting some delay while taking with software phone. I am
using Xlite software phone in both side. Please help me to reduce
this delay.
Regards
Niru
Drew Gibson wrote:
but ... why?
so windows lawyers can sneak a few patents thru the patent office
and sue Digium for patent infringement.
I am not criticizing Zoa or Luigi here, just reflecting on what ends up
happening eventually.
Think BSD code into windows, think file
change the caller id display for inbound calls and still have the
directory work properly?
Thanks in advance,
Max
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api
1 active SIP channel
===
There is a vtun IP tunnel between the Call routing asterisk server and
the Client asterisk server (the 10.3.0.0/24 subnet.) The 10.0.0.0/24
subnet is the client's LAN.
Any tips / ideas on what to try next are appreciated.
- Max
Max Bergmann schrieb:
How can i programming a Cisco 7961 to be used as busy lamp field?
my configs :
sccp.conf :
[devices]
type= 7961
tzoffset= 0
autologin = 601
speeddial = *31, Hanna -- other SIP telefon
extensions.conf :
exten = *31,hint,SIP/hanna
exten = *34,hint,SCCP
How can i programming a Cisco 7961 to be used as busy lamp field?
my configs :
sccp.conf :
[devices]
type= 7961
tzoffset= 0
autologin = 601
speeddial = *31, Hanna -- other SIP telefon
extensions.conf :
exten = *31,hint,SIP/hanna
exten = *34,hint,SCCP/601
on SIP Telefon (
Hi all,
Asterisk 1.4 was originally scheduled to be released early July
2006. Is there an update on the expected release of this version?
Also is there a changelog or feature list available that lists the
differences over 1.2?
TIA,
Max
--
Max Clark
http://www.clarksys.com
in the two
following e-mails. Use the function mysaynumber instead of
calling the say_number AGI function.
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web: http://www.comtel-networks.com
USA
Phone: 1 (877) 467-2877
ext. 1011001
Fax: (954) 827-0990
Venezuela
Teléfono: (0500) MAXITEL
ext
(digits/$sound_map/100-and);
$AGI-verbose(
Hundred and sound file: 100-and, $verbose );
}
}
The rest of the function continues on the third e-mail
With best regards,
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web
eq
0;
}
$AGI-verbose( RES: $res, $verbose )
if ( $config{debug_agi} eq YES);
$res = sprintf(%c, $res) if ( length(
$res ) );
return $res;
}
Hope this helps someone as it worked for me.
With best regards,
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web
please guide me to recompile after making the modifications?
I’d be happy to publish whatever I come up with; it doesn’t really seem to be
too complicated but it has been a very long time since I compiled my last C
program ☺
Your help will be greatly appreciated.
With best regards,
Max
,
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web: http://www.comtel-networks.com
Venezuela
Teléfono: (0500) MAXITEL
ext. 1011001
Fax: (0212) 953-0769
USA
Phone: 1 (877) 467-2877 ext. 1011001
Fax: (954) 827-0990
Comtel Networks, Corp. - Proprietary and
Confidential
BEGIN:VCARD
Hello,
Anyone knows a way to show real-time content from a DB into the LCD display
of an IP phone, like any 79xx?
If someone knows which phone is capable of doing and how, like using XML
files, please advise.
Regards,
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web: http://www.comtel-networks.com
Moreover, which phone can we use? We have a call shop cashier attended
feature for call shops, but still need to display the call to the booth
user...
Regards,
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web: http://www.comtel-networks.com
Venezuela
Teléfono: (0500) MAXITEL ext. 1011001
Fax: (0212
Thank You.
On 2/21/06, C F [EMAIL PROTECTED] wrote:
http://bugs.digium.com/view.php?id=5574
That is a patch that will do just that.
On 2/21/06, Max Clark [EMAIL PROTECTED] wrote:
Hi all,
I am interested in a follow me script for Asterisk - specifically I am
looking for one
- wrong password on authentication for INVITE error.
The problem is that setting fromuser in the sip.conf overrides
anything that I have set in the dialplan with SetCallerID. How do I
work around this?
TIA,
Max
--
Max Clark
http://www.clarksys.com
angle.
Resuming, we need to find support to modify rtp.c or dsp.c in order to
silence audio when tones are sent (received in *) from the user to * through
providers using CODECS G.723 and G.721 and DTMF recognition method RFC2833.
Regards,
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web: http
.
Is there anything like this available as an example for Asterisk?
TIA,
Max
--
Max Clark
http://www.clarksys.com
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
), or switch to a service with static registration that can be protected with a good firewall.
Max
On 1/23/06, JCC [EMAIL PROTECTED] wrote:
I've had problems for the last couple of weeks regarding incoming calls. Cant hear the party calling me (their voice sounds garbled/scrambled). If you haven't done
1 - 100 of 241 matches
Mail list logo
