Dear friends,
I am trying to get rxfax and txfax in my Asterisk 1.4.18 with no luck.
Everytime I get to the point Asterisk tries to compile the app_rxfax.c and
app_txfax.c I downloaded from agx-addons, the Asterisk make process crashes
with error. I checked all the doc I can find from google but
Is the Asterisk server yours? I am trying to figure out if Asterisk is in
your control and if it could be a problem at Asterisk, rather than your
SJPhone or your script, because I don't see any glaring problems in the
script.
Regards,
Amit.
-Original Message-
From: [EMAIL PROTECTED]
On Thu, Apr 03, 2008 at 06:06:28PM -0600, Al lists wrote:
you can see users status in Jaber,
Install Open fire Jabber server with Asterisk pluging.
And install one specific Jabber client, IIRC.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406
Hi,
I am using spandsp-0.0.4, tiff-3.8.2, and agx-ags-addon with Asterisk
1.4.18.
Everytime rxfax executes, Asterisk crashes:
-- Executing [EMAIL PROTECTED]:1] Set(Zap/2-1,
FAXFILE=/var/spool/asterisk-fax/1207322398.0.tif) in new stack
-- Executing [EMAIL PROTECTED]:2] RxFAX(Zap/2-1,
Yes, we mostly talk about asterisk, hardware, software, phones,
people, events and things asterisk-related. Asterisk is a registered
trademark of Digium. Asterix is a registered trademark of René
Goscinny. We never discuss that, though.
Here's the short URL for sending out to others that might be
Hi,
Could someone please help me with this?
I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on
an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS,
where an incoming line is plugged and also analog phone plugged to the
FXS port. Am faced with the problems
Hi All,
i´ve dealed around with two E400P Cards the last 3 Days and I cant
figure out why this problem exists.
Here the description:
I have two E400P Quad E1 Cards. One is in NT Mode and the other is in
TE(CPE) Mode.
When I connect both cards via E1-Cross Cabel the cards wont sync. I see
in
On Fri, Apr 04, 2008 at 11:52:33AM +0200, Bernhard Wessels wrote:
Hi All,
i´ve dealed around with two E400P Cards the last 3 Days and I cant
figure out why this problem exists.
Here the description:
I have two E400P Quad E1 Cards. One is in NT Mode and the other is in
TE(CPE) Mode.
Just my guess:
None of the spans (ports) is configured to provide clock to the other
side.
If you want to cross-connect port 1 and port 2, you should use something
like:
span=1,1,4,ccs,hdb3,crc4
span=2,0,4,ccs,hdb3,crc4
(And run ztcfg after that)
Yes i know. The NT System is providing the
Hi,
Your question is appropriate because you are asking about the best
design approach. Although it does have economic issues related.
First, let us dispatch the economics because they will impact your
technical approach. The real issue here is not cost of additional E1
but exposure to
Thinking out loud: write a asterisk call file (when the calling user
presses 5) which keeps on trying to connect the two.
On Fri, 2008-04-04 at 10:35 +, Tony Mountifield wrote:
Has anyone here implemented Ring back when free in Asterisk?
The way it works in the UK is as follows:
1. A
Has anyone here implemented Ring back when free in Asterisk?
The way it works in the UK is as follows:
1. A calls B. B is engaged (busy).
2. A hears The number you called is busy. To use ringback, press 5
3. A presses 5, and hears Your ringback request has been accepted.
4. A hangs up.
5. Later,
I posted this to asterisk biz but didn't get a reply.. I didn't want to
offend anyone being that this is kind of branching into hosting, and
maybe outside of the remit of this list.
Hi
Been lurking on the user list for a while but I have some what of an
immediate requirement and I'm wondering if
On 10:35, Fri 04 Apr 08, Tony Mountifield wrote:
Has anyone here implemented Ring back when free in Asterisk?
The way it works in the UK is as follows:
1. A calls B. B is engaged (busy).
2. A hears The number you called is busy. To use ringback, press 5
3. A presses 5, and hears Your
Seems to me that someting is wrong in layer1 setup. dont know why both
e400p cant sync. but they sync with rhino card?!?!
Strange.
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Hi,
I have a disk crash on an 2006 vintage Asterisk box that has a g729
license from Digium.
I have been able to re-install from media on the same chassis.
Reactivation is in progress.
Good so far. . .
HOWEVER, I cannot locate my g729 files for the 'digits' portion of
the sounds.
I only need
Wow! Fast response. Thanks. I am back up now. ..mike..
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On Fri, Apr 04, 2008 at 07:14:03AM -0400, Mike Trest - Personal wrote:
Hi,
I have a disk crash on an 2006 vintage Asterisk box that has a g729
license from Digium.
I have been able to re-install from media on the same chassis.
Reactivation is in
Hey Tony, in the pbx space that feature is called camp on
I have no idea why - just thought I'd let you know.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED]
Interesting to note that Tokbox now has 'clientless' voice and video
conferencing in the browser.
Does anyone know how they do this? Any thoughts on how we can leverage
off this for the asterisk community.
http://tokbox.com/
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
Jerry Geis wrote:
ounds like the module got auto-unloaded due to not being in use.
I have found the most reliable way to invoke zaptel/ztdummy is using the
proper init script:
1. In your zaptel source directory, do make config. That will create
/etc/rc.d/init.d/zaptel and the rcX.d links
Hi
I have two users on one asterisk server
A(192.168.1.100)
while the other two users are registered with asterisk
server B(192.168.2.50) and i have configured a PC to
act as a router between two subnets(Ping and other
applications are working). How i should configure
those two asterisk servers
Looks like a standard chatbox with flash media server in between.
You can't use this with asterisk unless you write a flash media server
channel or a convertor of some kind.
Zoa
Dean Collins wrote:
Interesting to note that Tokbox now has ‘clientless’ voice and video
conferencing in the
Hi all,
I need some sample configuration files (in xml format) for some of my atas,
spa-2102, 1001, 2002, 3000. If anybody can provide these, i'll be very glad.
I have heard that some people can retrieve the configuration file from the
ata. I have all the above mentioned ata's, so if you can
core show application retrydial
You need to do a core show applications and look at what apps are
included with Asterisk.
Tony Mountifield wrote:
Has anyone here implemented Ring back when free in Asterisk?
The way it works in the UK is as follows:
1. A calls B. B is engaged (busy).
2.
In article [EMAIL PROTECTED],
Faraz R. Khan [EMAIL PROTECTED] wrote:
Thinking out loud: write a asterisk call file (when the calling user
presses 5) which keeps on trying to connect the two.
I thought about that, but the trouble is, it's not event-driven. It just
keeps on trying until it runs
In article [EMAIL PROTECTED],
Jerry Geis [EMAIL PROTECTED] wrote:
Well after conversations on the list (thanks so much for responses) and
much google searching
it comes down to adding a kernel boot command line of acpi=off to
grub.conf.
This keeps the real time clock ticking and playing
In article [EMAIL PROTECTED],
Michiel van Baak [EMAIL PROTECTED] wrote:
On 10:35, Fri 04 Apr 08, Tony Mountifield wrote:
Has anyone here implemented Ring back when free in Asterisk?
The way it works in the UK is as follows:
1. A calls B. B is engaged (busy).
2. A hears The number
In article [EMAIL PROTECTED],
Dean Collins [EMAIL PROTECTED] wrote:
Hey Tony, in the pbx space that feature is called camp on
I have no idea why - just thought I'd let you know.
Thanks Dean - I've heard the term camp on, but it's seldom used in the UK,
and I thought it meant something more
In article [EMAIL PROTECTED], Eric Wieling [EMAIL PROTECTED] wrote:
core show application retrydial
But that doesn't allow the caller to hang up and get called back. It just
retries while he waits on the line.
You need to do a core show applications and look at what apps are
included with
On Tue, Apr 1, 2008 at 11:03 AM, Mike Trest - Personal [EMAIL PROTECTED]
wrote:
At 01:13 PM 4/1/2008, you wrote:
Is app_conference stable now?
I've never made it through a thousand calls without a crash. (With a
busy call center this doesn't take all that long.)
I have deployed a
Trouble with making it event driven (through the 'h' extension) is as
follows:
Imagine the user simply has his phone OFF hook (analog phone or an ip
phone like the grandstream which would report BUSY if cradle is
lifted).
Placing this cradle back generates NO sip events, the phone simply
Hi, I have released 0.28 the other day... I will probably make a new
branch for asterisk 1.6 soon..
On Thu, Apr 3, 2008 at 3:42 PM, Dean Collins [EMAIL PROTECTED] wrote:
Cute :)
I was thinking about getting something more complex developed but yes
FOP is a great product though getting a
As other people mentioned, there is also a DHTML client, if you look
at the javascript code, it is not hard to pull your own interface for
it... like:
1) Extension list like a grid, you can manage it via keyword only (hit
F for filter, type some letters to narrow the list,, hit T for
transfer
Faraz R. Khan wrote:
One of our clients is using a Grandstream GXP2000 with an attendant
console. We have used the same phone with past clients successfully
however this particular operator processes around 200 calls a hours and
the GXP2000 for sure does not like the quick line shuffling and
Hello Ivan,
I don't see nothing wrong in terms of signaling. When your side
(Asterisk/Unicall) request ANI, the other end answer with the signal
F, which means No More ANI, hence you receive an empty ANI string.
When your side request DNIS, the other end does not answer in several
seconds, which
On Thu, Apr 3, 2008 at 12:16 PM, Rajkumar S [EMAIL PROTECTED] wrote:
If some one has a combined patch that addresses both this issues for 1.4.x
series
that would be great!
Just caughtup with Atis in #asterisk and got the url of
state_interface patch against 1.4.19, its at
: Expression result is '0'
Apr 4 17:49:14 DEBUG[6271] app_macro.c: Executed application: GotoIf
Apr 4 17:49:14 DEBUG[6271] pbx.c: Function result is '20080404-174914'
Apr 4 17:49:14 VERBOSE[6271] logger.c: recordingcheck|
20080404-174914|1207320554.10: Outbound recording not enabled
Apr 4 17:49:14
No. I dont manage the asterisk server. I just manage my
proxy/firewall, and i need to my users can connect to that server.
The asterisk server is of another company. But the laptop users can
connect without problem to that server, from any place, except from my
LAN.
This is my scenarie:
Hi,
Returning to my office I find two missed calls (from autodialers) that
my IP601 displays as originating from 011. However the CDR
database recorded the call this way:
calldate: 2008-04-04 14:18:16+02
clid: 0172752780
src:
Hi list,
I enabled the transfer function in my dialplan, but I may configure it
wrong because sometime when I call a SIP extension number from one FXS
port, the SIP side will hear word transfer, I hear nothing, after
that, the call conversation is fine.I'v had this problem for a long
time, could
Hi,
I re-created your setup locally - even ran your script to setup NAT
192.168.12.2 (SJPhone) 192.168.12.1 (NAT-1)
|
-(nat box)
|
(NAT-1) 10.0.15.101 --- 10.0.15.102 Asterisk
And another client (Xlite) behind a
Your problems reflect what grandstream phones do.
I have been using a Polycom 650 with 2 sidecars as well as 601 with no
problems, we don't do paging, but the 650 does around 300 calls an
hour. With no problems.
Asterisk 1.2.x does support BLF. Earlier versions of 1.2.x would lose
subscriptions
Hi all,
Here's our setup:
Asterisk 1.4.18
Agx-ast-addons 1.4.5
Problem:
When accepting a fax, the fax itself comes through just fine, and it
does successfully create a tiff file. However, the dialplan should be
executing a system command right after that completes, but isn't due to
hanging up
We have occasional problems with failed transfers. The PSTN call comes
into Cisco Call Manager, then to asterisk over a SIP trunk and then to
an asterisk controlled SIP phone. The SIP phone transfers back to a
CallManager controlled SCCP phone which sometimes fails.
Is there a wait to let
Rob Schall wrote:
Hi all,
Here's our setup:
Asterisk 1.4.18
Agx-ast-addons 1.4.5
Problem:
When accepting a fax, the fax itself comes through just fine, and it
does successfully create a tiff file. However, the dialplan should be
executing a system command right after that completes, but
Hello everyone.
I wish I could continue with the approval of the engine Lumenvox, for
voice recognition, but not a development of acceptable engine,
Please could help me in achieving test?
As I said earlier we have a project that will involve a very large
number of licenses for Voice recognition,
Still no luck..
I still get this:
-- Accepting call from '3126290600' to '3125727758' on channel 0/1,
span 2
-- Executing [EMAIL PROTECTED]:1] Macro(Zap/25-1,
faxreceive|7758|[EMAIL PROTECTED]) in new stack
-- Executing [EMAIL PROTECTED]:1] Answer(Zap/25-1, ) in new stack
--
Rob Schall wrote:
exten = s,n,Hangup()
exten = h,1,System(/usr/bin/mailfax ${ARG1} ${CALLERID(num)}
${CALLERID(name)} ${FAXFILE}
Can you and a NoOP and send the output:
exten = h,n,NoOP(/usr/bin/mailfax ${ARG1} ${CALLERID(num)}
${CALLERID(name)} ${FAXFILE})
Doug
--
Ben Franklin quote:
Still no luck,
-- Accepting call from '3126290600' to '3125727758' on channel 0/1,
span 2
-- Executing [EMAIL PROTECTED]:1] Macro(Zap/25-1,
faxreceive|7758|[EMAIL PROTECTED]) in new stack
-- Executing [EMAIL PROTECTED]:1] Answer(Zap/25-1, ) in new stack
-- Executing [EMAIL
Asterisk Users,
I am running Asterisk 1.4.11, Zaptel
1.4.5.1, and Librpi 1.4.1 on a Debian Etch system. On the recorded
voice mail messages, the volume is really low when retrieving them with
my cell phone. I tried with multiple cell phones with the volume level
high and still, the same
Rob Schall wrote:
exten = h,1,System(/usr/bin/mailfax ${ARG1} ${CALLERID(num)}
${CALLERID(name)} ${FAXFILE}
Take the h extension out of the macro.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty
I'm set up to call 3 digit extensions at the office ( running 1.4.13)
from home ( 1.6.0 beta7) over iax. 1 out of 3 times the call breaks up,
but only in the home - office direction. office - home always sounds good.
If it were a poor internet connection, I'd expect both sides of the
Doug Lytle wrote:
Rob Schall wrote:
exten = h,1,System(/usr/bin/mailfax ${ARG1} ${CALLERID(num)}
${CALLERID(name)} ${FAXFILE}
Take the h extension out of the macro.
Doug
The problem is that I need the macro so I can make it mobile. I reuse
the macro about 20 times, and
Does this announce only on the user's side or the other side as well?
I mean if a user is on call with a customer, and you do the system announce,
will the announcement be heard by the customer as well? As that would be
bad, heh.
However, I'm really interested in checking it out. So, please
John Beaman
Telecom Specialist II
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 4/4/2008 3:23:49 PM
I'm set up to call 3 digit extensions at the office ( running 1.4.13)
from home ( 1.6.0 beta7) over iax. 1 out of 3 times the call
Nevermind. I'm a dummy. That method does work, but I didn't have a
${email} blah. Long week...
Thanks again for the help,
Rob
Rob Schall wrote:
Doug Lytle wrote:
Rob Schall wrote:
exten = h,1,System(/usr/bin/mailfax ${ARG1} ${CALLERID(num)}
${CALLERID(name)} ${FAXFILE}
I'm running Connected to Asterisk 1.4.18-1.
We have a T1 PRI and a SIP trunk coming in from our IVR.
When people call in on the SIP trunk, I experience the following
problem:
I put the caller on hold, MOH starts. MOH quits after approx ten
seconds. When I pick the call back up off of hold, they
Does anyone know of a Trixbox like install that has the hylafax integration
rolled in?
Looking for basic fax to email support.
Thanks,
Thermal
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Or any fax solution?
On Fri, Apr 4, 2008 at 11:46 AM, Thermal Wetland [EMAIL PROTECTED]
wrote:
Does anyone know of a Trixbox like install that has the hylafax
integration rolled in?
Looking for basic fax to email support.
Thanks,
Thermal
___
elastix
Thermal Wetland wrote:
Or any fax solution?
On Fri, Apr 4, 2008 at 11:46 AM, Thermal Wetland
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Does anyone know of a Trixbox like install that has the hylafax
integration rolled in?
Looking for basic fax to email
James Finstrom wrote:
elastix
*shivers*
Lee.
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On Thu, 3 Apr 2008 22:30:10 -0500, kazabe [EMAIL PROTECTED] wrote:
I need connect some LAN stations with SJphone to an Asterisk Server
published on Internet. [...] I dont manage the asterisk server.
I just manage my proxy/firewall, and i need to my users can
connect to that server.
SIP works
hi folks.
i'm experimenting with iaxmodem + hylafax using DID to determine
where to send the fax to it's final destination. however i have
difficulties passing the DID information from iaxmodem to
hylafax.
in extensions.conf:
exten = _,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r)
exten =
faraz wrote:
We are working on something along those lines which we should be able to
release in a few months.
JFYI:
Gemeinschaft (http://www.amooma.de/gemeinschaft/) comes with a thing
called ast_extstated (it's in the sbin/ directory). It's a Perl
daemon listening to the Asterisk manager
On Fri, Apr 4, 2008 at 12:55 PM, James Finstrom
[EMAIL PROTECTED] wrote:
elastix
thank you james for the recommendation, I will check it out.
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John Beaman wrote:
John Beaman
Telecom Specialist II
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 4/4/2008 3:23:49 PM
I'm set up to call 3 digit extensions at the office ( running 1.4.13)
from home ( 1.6.0 beta7) over
I was quite surprised to find a message in my in box from SellVOIP a
day or two ago. It indicated I was running out of credit which was a
surprise as I thought they'd gone under a large number of months
back. So I ran upstairs, added their entry back to sip.conf,
uncommented a couple of lines
I use FreePBX for my dial plan. I have been using iaxmodem and hylafax for over
a year with DID routing to email and samba shares. This is our mail fax number
dial plan.
exten = +16516834019,1,Set(__FROM_DID=+16516834019)
exten = +16516834019,n,Gosub(app-blacklist-check,s,1)
exten =
I made some install scripts based on centos 4 or 5 like trixbox but without all
the junk. It does have some fax setup stuff in it that I use on our production
servers that's been working for over a year. I you need any help you can email
me directly.
Jonn
On Fri, Apr 4, 2008 at 10:19 PM, Ira [EMAIL PROTECTED] wrote:
I was quite surprised to find a message in my in box from SellVOIP a
day or two ago. It indicated I was running out of credit which was a
surprise as I thought they'd gone under a large number of months
back. So I ran upstairs,
On Fri, Apr 4, 2008 at 9:43 PM, sean darcy [EMAIL PROTECTED] wrote:
John Beaman wrote:
John Beaman
Telecom Specialist II
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 4/4/2008 3:23:49 PM
I'm set up to call 3
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