Good morning,
I have a simple question,
Is this problem would affect also an Asterisk 1.4.38 if Pedantic SIP
support: No in the Global Signalling Settings
For what I understood, no..
Or is it a simple way to postpone upgrade until next planned upgrade.
Best Regards
Le mardi 18 janvier 2011
I need in a strange applicatio a way to detect the tone (busy, ring
etc. etc.) of analog line (zap channel), while channel UP.
I found the application NV line detect, but is very old, and may be
not mantained.
I can patch asterisk to actually support this application but i think
someone other
Hi
Does anyone have any idea how long it will take for the new release of
asterisk 1.8 to make it to the digium yum repositories?
Thanks
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
_
--
Correcting the line to:
exten =
_0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,)
problem persists...
any other suggestions?
Best regards,
What does your trunkdial-failover-0.3 look like?
Here goes...
[macro-trunkdial-failover-0.3]
exten =
Hey guys,
I hope somebody has some experience with the following because i'm stuck
;-).
I'm creating a fail over situation for Asterisk and this works great. The
only issue i have so fair os the from ip.
I used the IP fix routing here -
Hello list,
what does this mean in the debug-log :
[Jan 19 15:11:04] DEBUG[1475] audiohook.c: Write factory 0x153cf678 was
pretty quick last time, waiting for them.
[Jan 19 15:11:04] DEBUG[1701] audiohook.c: Read factory 0x14fe5ef0 was
pretty quick last time, waiting for them.
[Jan 19
On Sun, Jan 16, 2011 at 9:47 PM, James Miller paramedi...@gmail.com wrote:
When you get over 500 emails a day on your blackberry you have make a
decision on what is or is not worth reading at that moment.
Its not lazy at all its cutting through the fluff and finding the emails
worth while.
Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for everything else)
Question:
how can I know if the call was not answered because of timeout or because the
callee was busy ?
(I'm using
Am 19.01.2011 16:57, schrieb mancyb...@gmail.com:
Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for everything else)
Question:
how can I know if the call was not
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten
Göllner
Sent: Wednesday, January 19, 2011 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] agi dial termination cause ?
On Wed, 19 Jan 2011 17:03:03 +0100
Thorsten Göllner t...@ovm-group.com wrote:
Am 19.01.2011 16:57, schrieb mancyb...@gmail.com:Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for
On 01/19/2011 12:18 AM, randulo wrote:
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
That would fall under Quirk's Exception: Intentionally invoking Godwin's Law to
attempt to kill a thread is rarely successful.
On 01/19/2011 04:41 AM, Ishfaq Malik wrote:
Hi
Does anyone have any idea how long it will take for the new release of
asterisk 1.8 to make it to the digium yum repositories?
Thanks
Ish
They've been there since yesterday afternoon. It's possible that you hit the
repository before the
On 01/19/2011 12:18 AM, randulo wrote:
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
11:39 Parker said
That would fall under Quirk's Exception: Intentionally invoking Godwin's
Law to attempt to kill a
We're trying to forward an incoming SIP call from voipfone (UK ITSP) that
originated from a UK landline back up a SIP trunk to the same ITSP and on to
another UK landline number.
UK Landline-voipfone-asterisk 1.4-voipfone-UK landline
About 1 in 3 times the call at the final landline is silent
John Taylor wrote:
[snip]
Where do we start working out what's going on? Other than that the
server is working well
John
could you please ilustrate a little bit more your scenario?, (if you
want, use fake IPs).
Note:
What's the exactly version number of your Asterisk box?
--
Jose P.
I recently upgraded my office server to 1.8 and since then I have very
bad voice quality when calling another Asterisk server that uses 1.6.
The links is via IAX2 and I have tried using g729 and ulaw but I still
have the same problem although ulaw has a slight better result.
Any
On Wed, Jan 19, 2011 at 6:47 PM, Don Kelly d...@donkelly.biz wrote:
11:39 Parker said
That would fall under Quirk's Exception: Intentionally invoking Godwin's
Law to attempt to kill a thread is rarely successful. :)
Didn't work this time :)
Slightly OT: why is the Gmail ad server, which is
On Wed, Jan 19, 2011 at 2:37 PM, randulo rand...@randulo.com wrote:
Slightly OT: why is the Gmail ad server, which is usually all about
PBX, Asterisk, etc, now showing me Justin Beiber concert tickets on
this thread? Are they seeing it as that childish?
/r
Also OT: Google combines message
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Deneen
Sent: Wednesday, January 19, 2011 1:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Top Posting
On
Hi Steve,
The asterisk CLI shows the context of caller as below:
*moment-portable*CLI sip show user IMSI310410270465840
moment-portable*CLI
* Name : IMSI310410270465840
Secret : Not set
MD5Secret: Not set
Context : sip-external
Language :
AMA flags:
On Wed, 19 Jan 2011, abhinav anand wrote:
The asterisk CLI shows the context of caller as below:
moment-portable*CLI sip show user IMSI310410270465840
Context : sip-external
But when I do dialplan show 2103@sip-external, it returns no dialplan
moment-portable*CLI dialplan show
On Wed, 19 Jan 2011, Steve Edwards wrote:
3) Do you start Asterisk with the ? command line option?
? = '-C'
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
Hi Steve,
Here are the answers to the questions.
*1) Do you need to do a 'dialplan reload?'*
I don't need to do a dialplan reload. Infact there is no such command as
dialplan reload. I simply do a reload each time I make a config change.
*2) Are you sure you are editing the extensions.conf that
Morning All,
My Google skills may be failing me as I can see people asking this but no
useful responses, I need a way to prioritise calls across queues - I can think
of ways to do this but they are far from elegant and this seems like such a
simple request I am sure I am missing something
Please do not add me or yourself to the address list. We should keep the
discussion on the list (and just the list) so it is available to everyone.
Also, top-posting is 'frowned upon.'
On Wed, 19 Jan 2011, abhinav anand wrote:
Here are the answers to the questions.
1) Do you need to do a
Hi Steve,
I am using Asterisk version 1.6.2.5-0. Surprisingly, I don't see dialplan
reload. When I do core show help dialplan I get list of commands as:
*
moment-portable*CLI core show help dialplan
dialplan debug Show fast extension pattern matching data
structures
Un-top-posting...
On Wed, 19 Jan 2011, abhinav anand wrote:
I am using Asterisk version 1.6.2.5-0. Surprisingly, I don't see
dialplan reload.
If you do not have 'dialplan reload,' you do not have pbx_config.so
loaded. Since pbx_config.so reads extensions.conf, if you don't have it
loaded,
On Wed, 2011-01-19 at 16:40 -0800, Steve Edwards wrote:
Un-top-posting...
On Wed, 19 Jan 2011, abhinav anand wrote:
I am using Asterisk version 1.6.2.5-0. Surprisingly, I don't see
dialplan reload.
If you do not have 'dialplan reload,' you do not have pbx_config.so
loaded. Since
Thanks Steve,
I figured out the problem. As you said correctly, *pbx_config.so* was not
getting loaded because in my extensions.conf file one extra file
extensions.local.conf was included which was actually not present in the
directory. I have commented that line and did *module load
We are setting up an office in Malaysia.
We contacted Telekom Malaysia and are surprised to be told that ISDN-30
is no longer available.
They are yet to give us information of the replacement technology.
Does anyone have any experience and information with this?
Thanks in advance.
--
On Wed, 19 Jan 2011, abhinav anand wrote:
I figured out the problem. As you said correctly, pbx_config.so was not
getting loaded because in my extensions.conf file one extra file
extensions.local.conf was included which was actually not present in
the directory. I have commented that line and
On 01/18/2011 08:17 PM, Shaun Ruffell wrote:
On 1/18/11 6:55 PM, sean darcy wrote:
On 01/18/2011 05:27 PM, Shaun Ruffell wrote:
On 01/18/2011 04:06 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hello Lee,
Telekom Malaysia provide PRI lines. We've been actively using their services
for the past few years with success. Let me know if you need contacts.
Regards,
Arstan
On Thu, Jan 20, 2011 at 9:56 AM, Lee, John (Sydney)
john@compuware.comwrote:
We are setting up an office in
Arstan, thank you for your response.
Malaysia Telekom replied This service is limited to avaibility of ports and
infra avaibility as we are now upgrading to NGN. You may use business broadband
and PSTN lines to connect to your Digital PABX to replace this service.
On Jan 19, 2011, at 3:18 PM, Jason Parker wrote:
On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
I am working on some fax tools for some of my users. I am reading the
https://wiki.asterisk.org docs for faxing.
Is see Application_SendFax and Application_SendeFax has one been
discondinued?
On Jan 19, 2011, at 10:06 AM, C F wrote:
On Sun, Jan 16, 2011 at 9:47 PM, James Miller paramedi...@gmail.com wrote:
When you get over 500 emails a day on your blackberry you have make a
decision on what is or is not worth reading at that moment.
Its not lazy at all its cutting through the
On Jan 19, 2011, at 5:06 AM, Vitor Carlos Flausino wrote:
In other words, which of the following is your situation:
1.) User dials 0X, asterisk sends 0X to the telco.
2.) User dials 0X, asterisk parses 0, strips it, and sends X
to the telco.
That might narrow it down.
I have an updated asterisk 1.8 server running on Freebsd 8.1, and
connecting through a Freebsd 8.1 pf firewall with a dumb modem adsl
connection (in other words FreeBSD is doing all the hard work). I am
trying to connect with Internode nodephone, but they aren't really
willing to spend the
There was a typo in the res_fax documentation. Application_SendeFax
should be the correct documentation. I don't know where Application_SendFax
is coming from - it's probably old. When the next import happens,
Application_SendFax should be replaced by the correct version (then I'll try
to
Hello,
I'm using AsteriskNow. Asterisk version is 1.6.2.15 and FreePBX 2.7.0.0
Is there anyway to play prerecorded agent intro-speech (like Hello, my name
is ) to outside caller when agent picks up?
thank you
--
_
--
Hi Lee,
yes, it depends on the location. Usually they will check the location to see
if it is available there. Do you have your location set already?
If you need help further help, we can take our conversation off the mailing
list.
Arstan
On Thu, Jan 20, 2011 at 11:14 AM, Lee, John (Sydney)
Hi all,
Can someone give me a direction on how to use asterisk and icecast or any
other apps for a live audio cast?
The audio feed is external to the asterisk server.
Voip-info.org is not detailed on this.
Thank you
--
_
--
Also OT: Google combines message context with your personal search
history to do ad targeting, so look in the mirror.
I just made that up, though.
Not your mirror - your cookies!
No, it's true! Now I'm seeing Untimate Black Hat SEO (yes misspelled
because Ultimate was too expensive)
I was
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