Has anyone sucessfully implimented a fax or modem detection dial plan? I'm
originating calls from asterisk using a list of numbers and dropping the
destination into an IVR menu but need to do something different if a modem or
fax answers. I tried to use the NVBackgroundDetect() application bu
For such a simple application I'd use AstDB to avoid having to hassle with
an external database (and also means this sort of dialplan will work even on
embedded/slimmed Asterisk boxes that may not have db modules
loaded/available). In any case, what Tilghman said is what I'd suggest as
well.
h
On Saturday 01 December 2007 18:09:27 Steve Johnson wrote:
> Hi List,
>
> We have a remote asterisk SIP phone at the cottage.
>
> I'd like it to have minimal privileges when it first registers with
> Asterisk. Ideally it should be in a restricted context. Dialing any
> number would intercept the c
On Thu, 29 Nov 2007 23:55:38 -0600, "John Faubion"
<[EMAIL PROTECTED]> wrote:
>The newer CF cards are making this nearly a mute point. Seems like I provide
>updated software often enough that I never have CF cards wear out.
I guess /tmp can live in RAM, but what about eg. recording ten-twenty
WAV
On Saturday 01 December 2007 09:43:41 equis software wrote:
> In Asterisk 1.4.15 if I try to configure cdr_pgsql.conf , asterisk crash
> with this message
>
> asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_pgsql.so:
> undefined symbol: PQescapeStringConn
>
> Is this a knowed error?
T
Steve,
You might be able to swing it using the configuration updater that's part
of the manager API as of 1.4.0:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+UpdateConfig
-- Alex
On Sat, 1 Dec 2007, Steve Johnson wrote:
> Hi List,
>
> We have a remote asterisk SIP
Avaya makes 52% of it's revenue from professional services. In enterprises,
you generally have 3 budgets: Captial, expense, & professional services
Avaya figured out that they could make more money tapping into professional
services portion of the budget with "charge by the hour" union consultant
Hi List,
We have a remote asterisk SIP phone at the cottage.
I'd like it to have minimal privileges when it first registers with
Asterisk. Ideally it should be in a restricted context. Dialing any
number would intercept the call and tell the person to log on. This
way, if the phone was stolen o
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Probably you have deny=something instead of disallow=all.
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (D
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equis software wrote:
> 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??
Check the directory you were in when you ran Asterisk.
- --
Kind Regards,
Matt Riddell
D
> [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
>> subsequent calls will be rejected.
>>
>> ___
>> --Bandwidth and Colocation Provided
Anthony Francis wrote:
> Philip Prindeville wrote:
>
>> 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
When you take your packet capture, you'll need to look at the sip messages
with SDP attached to get the ip's and ports used for both media streams.
Make sure that the ips are correct and that the port used can traverse
between those ip's without being blocked by a packet filter or firewall. A
lot o
Andrew Kohlsmith wrote:
> On Friday 30 November 2007 04:17:36 Philipp Kempgen wrote:
>> With SIP you can "attach" custom variables to calls (using
>> X-... headers).
>> IAX (Inter-Asterisk eXchange!) can't do that (yet).
>
> With IAX2 you can share variables too. I believe Tilghman had supplied a
On Friday 30 November 2007 04:17:36 Philipp Kempgen wrote:
> With SIP you can "attach" custom variables to calls (using
> X-... headers).
> IAX (Inter-Asterisk eXchange!) can't do that (yet).
With IAX2 you can share variables too. I believe Tilghman had supplied a
patch to do exactly that severa
bilal ghayyad wrote:
> Hi List;
>
> Anyone knows a method (command) to increase the voice
> volume at diguim card level?
Are you trying to do this at some other level than rxgain and txgain
settings in zapata.conf?
If so, for the analog cards there are some module parameters for doing
so. For
Fabiano Sidler wrote:
> Hello? Nobody any idea how to fix that or how to compile asterisk-1.4 for
> Debian/slug on an i686 machine? Or should i ask on the nslu2 mailing list?
You did not make clear if you try to build on an i686 or on
a slug (as your subject says) which is not x86 but Intel
XScal
On Fri, 30 Nov 2007 10:54:47 -0900, "Mojo with Horan & Company, LLC"
<[EMAIL PROTECTED]> wrote:
>Sorry it's in some pseudocode that doesn't really represent a language
>at all.
BTW, how do most people write dialplans these days? Do they still use
extensions.conf, or did they move to either AEL, A
On Nov 28, 2007, at 8:24 PM, [EMAIL PROTECTED]
wrote:
>> From memory - 'rtcachefriends=yes' should do the trick.
>
> PaulH
Sorry for the late response, wanted to make sure everything else was
still working. This did indeed solve the problem. The only side
affect I have noticed is that ch
Hi List;
Anyone knows a method (command) to increase the voice
volume at diguim card level?
Regards
Bilal
Be a better pen pal.
Text or chat with friends inside Yahoo! Mail. See how.
http://overview.ma
On Wednesday 28 November 2007 23:27:41 myself wrote:
> [..]
> dlfcn.c:1225: error: dereferencing pointer to incomplete type
> dlfcn.c:1225: error: dereferencing pointer to incomplete type
> dlfcn.c:1225: error: dereferencing pointer to incomplete type
> dlfcn.c:1225: error: dereferencing pointer to
Philip Prindeville wrote:
> 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
>>> "inco
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
>
>> Is /sbin in your path?
>>
>> CP
>>
>> Robert McNaught wrote:
>> >
>> > my problem is that a non-privileged u
[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
> subsequent calls will be rejected.
>
> ___
> --Bandwidth and Colocation Provided by http://www
steve,
oops, you are right... sorry.. wrong list...
daveC
Steve Edwards wrote:
On Sat, 1 Dec 2007, dave cantera wrote:
[snip]
You forgot "i don't know what the shift key is" and "i don't understand
what Non-Commercial Discussion means."
Thanks in advance,
---
In Asterisk 1.4.15 if I try to configure cdr_pgsql.conf , asterisk crash
with this message
asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_pgsql.so:
undefined symbol: PQescapeStringConn
Is this a knowed error?
___
--Bandwidth and Colocation
On Sat, 1 Dec 2007, dave cantera wrote:
[snip]
You forgot "i don't know what the shift key is" and "i don't understand
what Non-Commercial Discussion means."
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voi
Hi
after having installed asterisk 1.4.15 my voicemail does not work anymore.
Am I the only one?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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to all,
I am available for work either US or Non-US for * consulting,
configuring, integration with other business applications. have been
working with * for about three years on and off and would like to do
this full time. am available for on-site or remote project work.
have 20+ years UNIX
Do you mean only once per day?
On Dec 1, 2007 7:47 AM, Alex Balashov <[EMAIL PROTECTED]> wrote:
>
> 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 im
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.
>
Just look at the changelogs, there have been lots and lots and lots of
commits to the iax cha
Hi Veselin,
You can verify SDP and RTP by running protocol analyzer such Ethereal, if
you need instruction you can follow this link.
http://www2.cs.uh.edu/~jsteach/cosc4377/2000fall/ethereal.html
While TDM side, I think he is referring to the card, if the card is faulty
or not.
Best Regards,
Joa
Thanks !!
I am still confused, so are these functionalities SIPREFERREDBYHDR,
SIPREFERTO are included in the 1.4 version ?
- Arpit
On Dec 1, 2007 1:31 AM, Norman W. Franke <[EMAIL PROTECTED]> wrote:
> On Dec 1, 2007, at 12:30 AM, [EMAIL PROTECTED]
> wrote:
>
> > I would like to extract the inform
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 added
>
Am Freitag, den 30.11.2007, 15:08 -0800 schrieb Philip Prindeville:
> 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 "de
> Is the PCI slot large enough for full height, half length PCI boards ?
Yes.
> Has you heard of a PCI Express version ?
No but the way chipsets are coming down in price, I would imagine someone
will have it soon.
John
___
--Bandwidth and Colocatio
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