On Dec 1, 2007, at 12:30 AM, [EMAIL PROTECTED]
wrote:
> I would like to extract the information present in the SIP REFER
> message that comes to asterisk. Would SIP_HEADER() allow me to do that
> ? I have used SIP_HEADER() for extracting the to and from SIP headers
> previously.
I wanted to do
Salvatore Giudice wrote:
> They are cheap. You only have to pay for the box and the
> maintenance percentage.
That is indeed the Avaya way. First you buy it, then you rent it. Stop
paying their maintenance fees and their dial into your PBX and cripple
the OS by removing customer maintenance c
Anyone see an issue on asterisk 1.2 that it will not accept the invite
from a Cisco gateway. If I turn off voice service voip signaling
forward unconditional then Asterisk accepts the call but without cname.
Below is a trace.
Any help is appreciated.
Thanks
John Bittner
Simlab.net
voippbx01*
Hi,
I am using the OS which bundled with AsteriskNow
- Original Message -
From: Vivek Shrivastava
To: Newbie
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Saturday, December 01, 2007 12:25 PM
Subject: Re: [asterisk-users] Registration state: Failed
you can also look at this...
http://www.asteriskguru.com/tutorials/idefisk_20_free.html
"I has this error initially with Asterisk server when I try to register.
" Device does not match ACL "
got it resolved by setting Caller ID Name : " users exten "
On 11/30/07, Vivek Shrivastava <[EMAIL PR
Hmmm, what OS you are using,,,this could be related to "*Access Control
Lists"..*but i guess that is in Solaris * *
On 11/30/07, Newbie <[EMAIL PROTECTED]> wrote:
>
> Hello,
>
> After I turned on "full=>" in logged.conf .. I got the following:
>
> [Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Regist
Hi * users,
I would like to extract the information present in the SIP REFER
message that comes to asterisk. Would SIP_HEADER() allow me to do that
? I have used SIP_HEADER() for extracting the to and from SIP headers
previously.
Thanks
Regards
--
Arpit Mehta
Graduate Student
Department of Comp
Hello,
After I turned on "full=>" in logged.conf .. I got the following:
[Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Registration from 'FXS1 ' failed for '172.16.1.169' - Device does not match ACL
[Nov 30 12:00:29] NOTICE[2601] chan_sip.c: Registration from 'FXS1 ' failed for '172.16.1.169' - Devi
On Nov 29, 2007 5:49 AM, Mark Wiater <[EMAIL PROTECTED]> wrote:
> Russell Bryant wrote:
> > Ron McCarthy wrote:
> >> Asterisk 1.4 im guessing? I did not know the Snom's worked with that,
> >> Ill have to check it out then!
> >
> > The way it is implemented in Asterisk is a bit interesting. It use
well, then i would recommend to see "full" log in debug mode that might give
some clue. if you have not done this before you can uncomment line starting
with "full=>" in the logger.conf... the log will be the usual
/var/log/asterisk/ directory.
Thanks,
Vivek
On 11/30/07, Newbie <[EMAIL PROTECTE
Hi,
there is no problem with X-Lite, the problem is SPA-3102 shown:
Line 1:
Registration Status: Failed
PSTN Line 1:
Registration Status: Failed
I also had added 1 more extension 251..then tried to call 251 from 250 by using
X-Lite and it works perfectly.. so that's why I am sure there is no p
Hi,
x-lite has extensive debug facility you can turn that on in the advanced
options, that probably will give better understanding as what is going on
from x-lite side. i also have experienced the same but that involved
firewall and NAT issues.
Thanks,
Vivek
On 11/30/07, Newbie <[EMAIL PROTECT
Tilghman Lesher wrote:
> On Thursday 29 November 2007 13:29:17 Philip Prindeville wrote:
>
>> [snip]
>> The issue is that I have, per "virtual pbx" (i.e. home or business), two
>> contexts that these get used from. The "internal-xyzzy" and
>> "incoming-xyzzy" contexts (one for each pbx, ie. "xy
On Thursday 29 November 2007 13:29:17 Philip Prindeville wrote:
> I'm trying to set up my extensions.conf file using some of the existing
> macros like stdexten, etc. while at the same time having two logically
> separate virtual PBX's (with no "default" context) and two trunks coming
> into separa
On Friday 30 November 2007 17:33:09 Mojo with Horan & Company, LLC wrote:
> Tzafrir Cohen wrote:
> > On Wed, Nov 28, 2007 at 10:47:44AM -0900, Mojo with Horan & Company, LLC
wrote:
> >> You might want the directory structure at /var/lib/asterisk as well, as
> >> it contains the current state of th
Dear Support,
I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with
PSTN line.
I have 3 extensions:
250 -> my extension
998 -> I configured as Line 1 in SPA-3102
999 -> I configured as PSTN Line 1 in SPA-3102
I have created 998 and 999 to the user extension list of the
> "PK" == Philipp Kempgen <[EMAIL PROTECTED]> writes:
PK> Anthony Francis wrote:
>> 2. Many features such as hinting (BLF) do not work with
>> realtime.
PK> That's only true if *extensions.conf* comes from a db table.
Nope, turn off caching and use realtime for SIP peers, and suddenly
BLF do
Brian J. Murrell wrote:
> On Fri, 2007-11-30 at 15:08 -0800, Philip Prindeville wrote:
>
>> bump...
>>
>
> What's with all this "bump" I see here? Is this a web forum?
>
> b.
>
>
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-dig
On Fri, 2007-11-30 at 15:08 -0800, Philip Prindeville wrote:
> bump...
What's with all this "bump" I see here? Is this a web forum?
b.
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Thank you much for the prompt reply Salvatore.
Would you have the time to explain further how should I go for verifying
that SDP and RTP are OK.
Also what is reffered to as the TDM site.
Veselin
On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote:
> Take a packet capture of your Vo
Well, there you go then - either add /usr/sbin to your path, or provide
a full path thusly:
/usr/sbin/asterisk -r
CP
Robert McNaught wrote:
> not in path
>
> [EMAIL PROTECTED] echo $PATH
> /usr/kerberos/bin:/usr/lib/courier-imap/bin:/usr/local/bin:/bin:/usr/bin:/usr/X11R6/bin:/home/admin/bin
>
If the majority of the MoH is queues, move the location of the queue.
On Nov 28, 2007 4:42 PM, Darryl Dunkin <[EMAIL PROTECTED]> wrote:
> Does anyone have any opinions on the music on hold quality over G729?
> The stock files seem to sound terrible over it, this is enhanced further
> by calls comi
On Fri, 30 Nov 2007 00:30:06 -0500, Jared Smith <[EMAIL PROTECTED]>
wrote:
>Sounds like a perfect application for the ISNULL dialplan function. Of
>course, that adds a whole new set of curly braces and parentheses to
>watch out for.
Thanks Jared for the pointer :-)
exten => s,1,Set(foo=${ISNULL(
Mark Wiater wrote:
> I fought with this in 1.4.5 with polycom phones. I was hoping to share a DID
> from a PRI on several
> Polycom IP430's.
>
> Might you be willing to share some specific configurations for such a
> situation?
There are some basic examples in doc/sla.pdf in the 1.4 tree. Howe
On Fri, 30 Nov 2007, Robert McNaught wrote:
>> It seems that non-privileged users cannot run commands in sbin, but
>> can in bin directories
Unless something in your host is major league hosed, this is not true.
Try:
/sbin/runlevel
/usr/sbin/ntpdate -q 0.us.pool.ntp.org
Dependi
Steve Langstaff wrote:
> [line1_outbound]
> exten => disa,1,Disa(no-password|line1_outbound)
> exten => _,1,Dial(SIP/[EMAIL PROTECTED])
> exten => _,2,Hangup
> So to summarise:
> if I seize the line and dial a number known at vsp5000 then I
> get ringing etc - good.
> if I seiz
Griefs?
rejected connect attempt from 111.111.111.111, who was trying to reach
'12345678' No authority found
call rejected by 111.111.111.111: No authority found
But once it works it works...
I have DTMF issues with sending calls from 1.2 to what I suspect is a
really old 1.4 build via IAX that
Store a value indicating it has been called as a unique key in AstDB, and
set your dial plan to check for it.
On Fri, 30 Nov 2007, [EMAIL PROTECTED] wrote:
> Anyone have an idea how to implement a phone number that can only be
> called once? The first time it will process normally and any
> sub
not in path
[EMAIL PROTECTED] echo $PATH
/usr/kerberos/bin:/usr/lib/courier-imap/bin:/usr/local/bin:/bin:/usr/bin:/usr/X11R6/bin:/home/admin/bin
>
> Is /sbin in your path?
>
> CP
>
> Robert McNaught wrote:
> >
> > my problem is that a non-privileged user, eg admin, cannot log in and
> > conn
Anyone have an idea how to implement a phone number that can only be
called once? The first time it will process normally and any
subsequent calls will be rejected.
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asterisk-users m
Hi.
I am starting with asterisk, but I will not have problem to compile the
newer version 1.4.
My question is if it is worth to compile rather then using the binary
1.2 version in Debian stable?
I plan to use one analog PSTN line and two sip providers.
Thanks
Jiri
___
Todd wrote:
> Hi -
> I've got a new install with a Sangoma A200 and a few GXP2000's. When
> users are talking over the Sangoma, they get a lot of sidetone (local
> echo). Internal calls are fine. Where do I adjust that? I assume
> its in zapata.conf somewhere?
> thanks
> Todd
>
>
Dear The Expert,
I am very new with this, I have installed AsteriskNow, X-Lite as my
SoftPhone, I am using SPA-3102.
I have 3 extensions,
me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below)
My problem is, I am unable to call 998, I thought this is registration
problem, (bec
Hi there!
I am having problems registering my 7970 hardphone with Asterisk 1.4(with
FreePBX interface). I had an earlier post about trying to get it to work first
with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum :
http://forums.digium.com/viewtopic.php?t=19160
Instead I deci
Tzafrir Cohen wrote:
> On Wed, Nov 28, 2007 at 10:47:44AM -0900, Mojo with Horan & Company, LLC
> wrote:
>
>> You might want the directory structure at /var/lib/asterisk as well, as
>> it contains the current state of the voicemail boxes and any custom
>> sound files that might have been add
Version 33 of codec_g729a for Asterisk 1.4 has been released. This release is a
compatibility update to work with the latest version of Asterisk. Users of this
module upgrading to Asterisk 1.4.15 will need to upgrade to this version of
codec_g729a.
The module is available for download at the fol
Robert McNaught wrote:
>> thanks for the reply Tzafrir,
>>
>> I tried the below, but I think maybe I misexplained what I am trying
>> to do. I have asterisk running as user asterisk - I followed the
>> instructions in the Asterisk book and have everything stored in
>> /home/asterisk/asterisk-b
The Asterisk.org development team has released Asterisk-addons version 1.4.5.
This release contains a few bug fixes, but is required for compatibility with
the latest version of Asterisk, 1.4.15.
Thank you for your support!
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bump...
Philip Prindeville wrote:
> I'm trying to set up my extensions.conf file using some of the existing
> macros like stdexten, etc. while at the same time having two logically
> separate virtual PBX's (with no "default" context) and two trunks coming
> into separate contexts, i.e. one for res
Is /sbin in your path?
CP
Robert McNaught wrote:
>
> my problem is that a non-privileged user, eg admin, cannot log in and
> connect to the console by issuing the following
>
> [EMAIL PROTECTED] asterisk -r
> bash: asterisk: command not found
>
> [EMAIL PROTECTED] whereis asterisk
> asterisk:
> thanks for the reply Tzafrir,
>
> I tried the below, but I think maybe I misexplained what I am trying
> to do. I have asterisk running as user asterisk - I followed the
> instructions in the Asterisk book and have everything stored
> in /home/asterisk/asterisk-bin - this includes logs, pid fil
thanks for the reply Tzafrir,
I tried the below, but I think maybe I misexplained what I am trying to
do. I have asterisk running as user asterisk - I followed the
instructions in the Asterisk book and have everything stored
in /home/asterisk/asterisk-bin - this includes logs, pid files, configs
Take a packet capture of your VoIP segment and verify that the SDP is
correct and that the RTP is making it to the correct places. If all that
looks good and this is a straight out quality problem, then you need to
figure out if it's happening on the voip side or on the TDM side. You should
make ca
What if you set queue-thankyou to empty?
queue-thankyou = ""
I have a faint memory of doing this in the old 1.0 days... Not sure if it
works in the current releases...
// T
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
here's a do-while loop - the contents of the loop are executed BEFORE
the condition is tested.
--
labelA:
do some loopy things
if (we need to loop again)
goto labelA:
---
Hello,
I have an Asterisk running with a Sangoma A200 card with Hardware Echo
cancelling connected to the UK PSTN.
If a PSTN call comes in, voice both ways is OK, however if an outgoing
call over the PSTN is made I can hear the other party OK but they can
not, they can barely understand what I a
If you desire SIP in Avaya, you have to add a SES (SIP Enablement Server) to
your Avaya setup. They are cheap. You only have to pay for the box and the
maintenance percentage. You don't need to buy user ports or any of that
garbage as long as you setup your extensions using Optum, which is a free
A
Hi,
To make a long story short, I can't install any TAPI driver on my XP
platform.
A. Within Config Panel|Modems and Telephony options|Advanced parameters,
I've got a list of 7 TAPI drivers. Among them is Omniis TAPI driver for
Asterisk.
B. I can properly configure this driver (line, context, ...
I manage a large Avaya implementation with three systems at different
locations. I hate Avaya's manageability, lack of features, and
extremely high cost.
That's why I'm looking into alternatives to replace the whole thing in a
year or two.
I would appreciate any other opinions and findings r
Why not simply store voicemail local so there are no issues if the VPN
goes down. Then set up your dial plan at each site to allow the PSTN
access to your remote (other site) extensions. You can use IAX to trunk
a "PSTN" call just like you can a local caller, just give them access to
the context.
Hi,
I'm trying to set up a remote office with its own Asterisk Server they'll
have a dedicated land line, but we'll still want them connected to the main
office via VOIP (IAX2 via VPN). I've tested using IAX2 to bridge between
the offices based on extensions, since the extensions we want to share
It´s very strange, when Asterisk 1.4.15 crash don´t make a core file...
I´m sure it´s running with -g option!!
On Nov 30, 2007 11:02 AM, equis software <[EMAIL PROTECTED]> wrote:
> Hi, Jared. I'm going to test in 1.4.15 and then I'll tell you what
> happend.
>
> Thanks
>
>
> On Nov 29, 2007 3:3
Hello list,
I try to setup an asterisk-server with different SIP-Peers to PSTN.
The Peer are working and configured in sip.conf:
[peer1]
type=peer
host=10.10.10.1
[peer2]
type=peer
host=10.10.10.2
Now dialout is no problem. Extensions.conf says:
exten => _0Z.,1,Dial(SIP/49${EXTEN
Daryl G. Jurbala wrote:
> How recent? I tried switching from 1.2 to 1.4 about 4 months ago, and
> asterisk would stop accepting IAX connections in less than a day and
> would need to be restarted.
It has been a continuously worked on task (ever since a few months ago).
Russell Bryant and o
> > > > Short of replacing a sound file with a sound file containing only
> > > > a short period of silence, is there any way to suppress certain
> > > > sounds from playing during queue processing by configuring for
> > > > example queues.conf or other similar files?
> > >
> > > Which announcement
looks like something wrong with the dial plan in the extensions.conf.. i
would recommend start debug on and see the content of "full" log may be
that give some clue.
Thanks,
Vivek
On 11/30/07, Russell Brown <[EMAIL PROTECTED]> wrote:
>
>
> I have two Asterisk systems that can route to each oth
Hi!
> > > Short of replacing a sound file with a sound file containing only a
> > > short period of silence, is there any way to suppress certain sounds
> > > from playing during queue processing by configuring for example
> > > queues.conf or other similar files?
> >
> > Which announcements are
I have two Asterisk systems that can route to each other via a VPN with
firewalls disabled for testing purposes.
Each Server can see (tested via nmap) UDP port 5060 on the other.
So... I thought that I could simply use a Dial command in Server A's
config to place a SIP call to Server B... but
Sasa wrote:
> "Tzafrir Cohen" wrote:
>
>> New:
>> loadzone=it
>> defaultzone=it
>> span=1,1,3,ccs,ami
>> bchan=1,2
>> dchan=3
>> span=2,1,3,ccs,ami
>> bchan=4-6
>> dchan=6
>>
>>> ..in zapata.conf I have:
>> ; new part:
>> switchtype=euroisdn
>> signalling = bri_net
>> priindication=outofband
>> gr
You can try something like this:
exten => _X.,1,SET(condition=${RAND(1,2)})
exten => _X.,2,GotoIf($[${condition} = '1']?1:3)
exten => _X.,3,SET(Result is 2)
Regards,
Ricardo Carvalho.
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> [EMAIL PROTECTED] wrote:
> >> [EMAIL PROTECTED] wrote:
> >>> Short of replacing a sound file with a sound file containing only a
> >>> short period of silence, is there any way to suppress certain
> sounds
> >>> from playing during queue processing by configuring for example
> >>> queues.conf or
"Tzafrir Cohen" wrote:
> New:
> loadzone=it
> defaultzone=it
> span=1,1,3,ccs,ami
> bchan=1,2
> dchan=3
> span=2,1,3,ccs,ami
> bchan=4-6
> dchan=6
>
>>
>> ..in zapata.conf I have:
> ; new part:
> switchtype=euroisdn
> signalling = bri_net
> priindication=outofband
> group = 1
> channel => 1-2
> gr
[EMAIL PROTECTED] wrote:
>> [EMAIL PROTECTED] wrote:
>>> Short of replacing a sound file with a sound file containing only a
>>> short period of silence, is there any way to suppress certain sounds
>>> from playing during queue processing by configuring for example
>>> queues.conf or other similar
How recent? I tried switching from 1.2 to 1.4 about 4 months ago, and
asterisk would stop accepting IAX connections in less than a day and
would need to be restarted.
This is with about 50 to 100 calls at a time on each box for about 10
or 12 hours a day. Less for the other half. And all
Hi,
Is there a way to have a Do-While sort of loop, as opposed to a simple
While?
I have a condition that the loop depends on even for the first iteration, as
it often happens in life.
Regards,
Mike
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> [EMAIL PROTECTED] wrote:
> > Short of replacing a sound file with a sound file containing only a
> > short period of silence, is there any way to suppress certain sounds
> > from playing during queue processing by configuring for example
> > queues.conf or other similar files?
>
> Which announce
Hi, I'm testing Asterisk 1.4.15 with the -g option.
When it crash didn´t generate core file in the /tmp folder.
What is happening??
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This is both a hardware and software licensing issue.
Avaya offers a SIP server separate from their main VoIP gateway.
The core platform uses H.323.
Either SIP or H.323 has a license cost per registered device.
We have an Avaya S8300 Communications Manager providing H.323 and have this
tied to an A
[EMAIL PROTECTED] wrote:
> Short of replacing a sound file with a sound file containing only a
> short period of silence, is there any way to suppress certain sounds
> from playing during queue processing by configuring for example
> queues.conf or other similar files?
Which announcements are y
In zapata.conf you can add rxgain and txgain settings and use
ztmonitor to get it set. There are some more details on this on
voip-info.org.
On Nov 29, 2007 1:49 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Hi All;
>
> I have an IP Trunk established between Asterisk and
> the VoIP service provid
The jitter buffer is actually the same.
Zoa
Dr. Michael J. Chudobiak wrote:
> randulo wrote:
>
>> On Nov 30, 2007 1:40 PM, Steve Totaro <[EMAIL PROTECTED]> wrote:
>>
>>> solved these issues. I think trunking (one of the main selling points
>>> of IAX due to less overhead) may be a commo
The poll is still open here: http://food4wine.ning.com/poll
Here is a CSV file of the 99 answers.
http://voipusersconference.org/poll/
There is also an XML version, but it was created by Excel so I don't
know if it's worth dealing with:
http://voipusersconference.org/poll/results.xml
Because
Hi, Jared. I'm going to test in 1.4.15 and then I'll tell you what happend.
Thanks
On Nov 29, 2007 3:35 PM, equis software <[EMAIL PROTECTED]> wrote:
> You are right!
> Here there is the backtrace
>
> (gdb) bt
> #0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
> #1 0xb7cc4a3f in local_ast_mo
IAX had some stability issues in the past, the recent releases have a
lot of iax2 fixes and should no longer have those issues.
Zoa
Steve Totaro wrote:
> randulo wrote:
>
>> Hi,
>>
>> We all know what the principal advantage of IAX is, doing it all on a
>> single port, right? But now and ag
Same with me IAX trunking worked great up until about 10 calls. Then
it went down hill. This was back on 1.2, I haven't tried it since. So
maybe it has been fixed?
Doug
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of randulo
Sent: Friday, November 30
Dear all, sorry for my OT but I need to know if Avaya voip server uses
SIP or H.323 ???
Anybody can't tell me this...so I'm here for thei reason.
Thanks a lot
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asterisk-users maili
randulo wrote:
> On Nov 30, 2007 1:40 PM, Steve Totaro <[EMAIL PROTECTED]> wrote:
>> solved these issues. I think trunking (one of the main selling points
>> of IAX due to less overhead) may be a common denominator.
>
> That does tend to explain why I've never experienced (or at least
> noticed)
Short of replacing a sound file with a sound file containing only a short
period of silence, is there any way to suppress certain sounds from playing
during queue processing by configuring for example queues.conf or other
similar files?
___
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On Nov 30, 2007 1:40 PM, Steve Totaro <[EMAIL PROTECTED]> wrote:
> solved these issues. I think trunking (one of the main selling points
> of IAX due to less overhead) may be a common denominator.
That does tend to explain why I've never experienced (or at least
noticed) problems. I never trunk w
randulo wrote:
> Hi,
>
> We all know what the principal advantage of IAX is, doing it all on a
> single port, right? But now and again I hear complaints about it. What
> specific griefs have you had with IAX and has it stopped you from
> using it entirely? Under what conditions have you had problem
On 11/30/07, Vivek Shrivastava <[EMAIL PROTECTED]> wrote:
>
> you can try Cain & Abel ( to route calls) and Wireshark to record all the
> calls.
>
> On 11/29/07, Adam Moffett <[EMAIL PROTECTED]> wrote:
> >
> > I'm pretty sure asterisk won't do that without modification. You'll
> > need to do pack
On Wed, Nov 28, 2007 at 04:59:22PM +0100, Sasa wrote:
> Hi, sorry but perhaps I don't have explained clearly my problem...now I have
> a box voip that must be replace with another box voip but I want to do test
> before remove the old voip from production.
>
> The box voip (named 1) that now is
On Thu, Nov 29, 2007 at 11:14:12AM +0100, Sasa wrote:
> Hi, my problem isn't on new voip box with latest asterisk version...my
> problem is on voip with Asterisk 1.2.13 where I must remove TDM Card, this
> steps for remove rightly TDM Card:
>
> - remove line configuration about tdm card in zapat
Hi,
I'm trying to use latest versions of ActivaTSP and Asttapi with an
Astmanproxy-enabled 1.4 Asterisk.
Up to now, I can't find a way to teach Outlook 2002 how to use any of those
TAPI drivers: when using "Call this contact" in Outlook Contacts pane, I
can't see and select any TAPI driver.
Besid
"Tzafrir Cohen" wrote:
> You have been quite short on details. For instance: what distribution of
> Linux? What version of Zaptel?
>
> Do you have another Zaptel card? It seems you either have two zaphfc
> cards or one dual-BRI card. If so, the procedure is slightly more
> complicated, as you bas
randulo wrote:
> What
> specific griefs have you had with IAX and has it stopped you from
> using it entirely?
With SIP you can "attach" custom variables to calls (using
X-... headers).
IAX (Inter-Asterisk eXchange!) can't do that (yet).
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 12
yup with chan_oss
On 11/30/07, Olivier <[EMAIL PROTECTED]> wrote:
>
>
> 2007/11/30, Vivek Shrivastava <[EMAIL PROTECTED]>:
> >
> > I am not sure if this fits in your requirement but try "dial" command.
> >
>
> Do you mean, dialing both extensions one after the other and then, bridge
> them ?
> Or
email the biz list. you should get some one there.
- Original Message -
From: "Sahil Gupta" <[EMAIL PROTECTED]>
To:
Sent: Thursday, November 29, 2007 1:32 PM
Subject: [asterisk-users] Hylafax
> Hi,
> We seem to be having some teething issues with a new Hylafax - happy to
> pay
> someo
Hi,
We all know what the principal advantage of IAX is, doing it all on a
single port, right? But now and again I hear complaints about it. What
specific griefs have you had with IAX and has it stopped you from
using it entirely? Under what conditions have you had problems?
I have used SIP and IA
- Original Message -
From: "Tilghman Lesher" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, November 24, 2007 5:33 PM
Subject: Re: [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.
> On Saturday 24 November 2007 00:16:11 St
Brillant !!
I don't know why, I wanted to substitute extension keyword with a value.
Thanks for the tip.
2007/11/30, Philipp Kempgen <[EMAIL PROTECTED]>:
>
> Olivier wrote:
>
> > Usage2: originate extension [EMAIL PROTECTED]
>
> > I would like for example to call 0123456789 number from SIP/7530
2007/11/30, Vivek Shrivastava <[EMAIL PROTECTED]>:
>
> I am not sure if this fits in your requirement but try "dial" command.
>
Do you mean, dialing both extensions one after the other and then, bridge
them ?
Or do you mean using the asterisk Chan_OSS capabilities ?
Cheers
___
Olivier wrote:
> Usage2: originate extension [EMAIL PROTECTED]
> I would like for example to call 0123456789 number from SIP/7530 extension.
> My asterisk server is set to use "local" context for outgoing calls.
> My first idea was to type this :
> originate SIP 7530 [EMAIL PROTECTED]
How ab
Hi,
2007/11/30, John Faubion <[EMAIL PROTECTED]>:
>
> > Thanks for the tip. It seems like they no longer manufacture them:
> >
> > http://www.neoware.com/products/hardware/
>
> No, but the Neoware e140 has a PCI expansion slot, is expandable to 1GB
> RAM,
> and still has room inside the case for a
I am not sure if this fits in your requirement but try "dial" command.
--Vivek
On 11/29/07, Olivier <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I would like to originate my first call from CLI.
> As I'm new to this, I'm wondering if it's possible.
> When I type "originate" from CLI, I've got this :
>
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