Steve Totaro wrote:
Thomas Kenyon wrote:
Steve Totaro wrote:
No, I mean specifying the group in your zapata.conf file and then
changing your dial statement to dial out like Dial(ZAP/g0/${EXTEN})
the g0 refers to group zero. If specify a channel to belong to
group=1 then it will not be
On Sun, Jun 25, 2006 at 08:28:35PM +0100, Thomas Kenyon wrote:
I have a TDM400 card with 3x FXO and 1x FXS ports on it.
At the moment I'd prefer (till I can get it working more reliable with
iaxmodem), for a faxmodem to answer one of the lines instead of the
linecard.
I've tried changing
26 jun 2006 kl. 07.10 skrev Martin Joseph:
On Jun 25, 2006, at 4:11 PM, Jean-Michel Hiver wrote:
Hi List,
Is there a way to tell asterisk to only accept SIP streams from
the same IP address that is used for signaling?
SIP streams are signalling... Have you tested the ACL features in
On Mon, Jun 26, 2006 at 07:37:00AM +0100, Thomas Kenyon wrote:
Steve Totaro wrote:
Thomas Kenyon wrote:
Steve Totaro wrote:
No, I mean specifying the group in your zapata.conf file and then
changing your dial statement to dial out like Dial(ZAP/g0/${EXTEN})
the g0 refers to group
Hi all,
first of all thanks for your comments and ideas. I wrote because i
wanted to know if i'm wrong or not and to let others khow how some
companies operate.
I work on technology, i work in the world we move, and i usually are
in charged of handle situatios like that, and what i can tell all
On Friday 23 June 2006 11:07, Kevin P. Fleming wrote:
- Dmytro Mishchenko [EMAIL PROTECTED] wrote:
In this case during all conversation SIP packets contains
Call-ID: [EMAIL PROTECTED]
but the final BYE packet from adapter contains
Call-ID: [EMAIL PROTECTED]
Is such scenario
Can the TE406P card's VPM module be swapped for the new revision with
Octasic chipset?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Sunday, June 25, 2006 8:08 PM
To: Asterisk Users Mailing List -
Johansson Olle E a écrit :
26 jun 2006 kl. 07.10 skrev Martin Joseph:
On Jun 25, 2006, at 4:11 PM, Jean-Michel Hiver wrote:
Hi List,
Is there a way to tell asterisk to only accept SIP streams from the
same IP address that is used for signaling?
SIP streams are signalling...
Sorry,
Well now would be a great time to come back, Doug! We miss you! 8)
--Rob
-Original Message-
From: Douglas Garstang [mailto:[EMAIL PROTECTED]
Sent: Monday, 26 June 2006 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
Asterisk Users Mailing List -
On 06/25/06 19:01 Roy Sigurd Karlsbakk said the following:
seem to pass all SIP and RTP traffic through their own servers... See
http://karlsbakk.net/asterisk/gizmo-project.php for details
interesting. but isnt Gizmo an open source client ?
--
Regards, /\_/\ All
If an agent dumps the call during an announcement (the immortal line in
app_queue.c is Agent on %s hungup on the customer. They're going to be
pissed) is there anyway of tracking this via a variable or hangupstatus
or something - I need to be able to trap this in the dialplan
Julian.
Hi All. Somebody works with asterisk linked in ISDN PRI with protocol QSIG with some PABX as Siemens, Philips, etc. The applications as pickup between asterisk and the PABX function? The names in the display and the number of the origin also? Which features that they can be used between the
Hello all.
I have installed and functioning asterisk-1.2.9.1 where I effected one upgradein asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destinedtoSIP phones
Hi Kevin.
Where could I get more information about those boards?
Thanks,
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1610A
CEP: 80530-000 - Curitiba - PR
+55 41 3252-2977 r 101
http://www.isolve.com.br
On 25 de jun de 2006, at 07:07, Kevin P. Fleming
Thomas Kenyon wrote:
I have a TDM400 card with 3x FXO and 1x FXS ports on it.
At the moment I'd prefer (till I can get it working more reliable with
iaxmodem), for a faxmodem to answer one of the lines instead of the
linecard.
I've tried changing the context of that line so that the exten = s
I agree whole-heartedly. If I could run this on my dedicated Asterisk
machine it would be perfect...
On 6/28/06, Matthias Fechner [EMAIL PROTECTED] wrote:
Hi Marco,
Marco Mouta wrote:
Please feel free to contact me if you have more ideas to improve this
solution, currently i didn't test
Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo?
É freeware?
Podemos seguir com o projeto Asterisk-PT?
Saudações
Josué
2006/6/26, John Klimek [EMAIL PROTECTED]:
I agree whole-heartedly.If I could run this on my dedicated Asteriskmachine it would be
On Mon, Jun 26, 2006 at 12:08:48AM -0400, Doug Crompton wrote:
Still awfully pricey for home use and the styling is not there for a
bedroom or many other areas of a modern home. What we need is a wireless
sip phone modeled like the panasonic or uniden which allow multiple
extension off of one
Bom dia,
On 6/26/06, Josué Conti [EMAIL PROTECTED] wrote:
Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
externo?
Sim é um software da Uplink, disponível para download gratuitamente, n
garanto q seja freeware (talvez tenha limitações esta versao free)
What on earth is going on with the list?!?! Some of my messages
never make it... then days later I get something like this back:
Final-Recipient: rfc822; asterisk-users@lists.digium.com
Action: failed
Status: 5.0.0
Diagnostic-Code: X-Postfix; mail forwarding loop for
Our main office is near Lansing, but we have a person who lives in the
AA are that would like to attend such a group.
On Thu, Jun 22, 2006 at 04:27:02PM -0400, BerkHolz, Steven wrote:
I am thinking of getting an asterisk user group together for either SE
Michigan or just Metro-Detroit.
How
OK Marco, irei efetuar os testes.
Se você quiser, posso lhe ajudar no forum, estou a disposição.
Assim que você criar as contas avise para podermos já ir colaborando.
Saudações
Josué
2006/6/26, Marco Mouta [EMAIL PROTECTED]:
Bom dia,On 6/26/06, Josué Conti [EMAIL PROTECTED]
wrote: Marco, bom
Matt wrote:
What on earth is going on with the list?!?! Some of my messages
never make it... then days later I get something like this back:
Final-Recipient: rfc822; asterisk-users@lists.digium.com
Action: failed
Status: 5.0.0
Diagnostic-Code: X-Postfix; mail forwarding loop for
On Mon, Jun 26, 2006 at 06:06:01PM +0800, Dinesh Nair wrote:
On 06/25/06 19:01 Roy Sigurd Karlsbakk said the following:
seem to pass all SIP and RTP traffic through their own servers... See
http://karlsbakk.net/asterisk/gizmo-project.php for details
interesting. but isnt Gizmo an open
Hi!
I'am often see this WARNINGs in messages file. What does it mean?
Jun 26 16:59:00 WARNING[62792] chan_sip.c: Insufficient information for SDP (m
= '', c = '')
Jun 26 16:59:01 WARNING[62792] chan_sip.c: Insufficient information for SDP (m
= '', c = '')
And it seems that at this time I
On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:
Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
externo?
É freeware?
Podemos seguir com o projeto Asterisk-PT?
English, please, folks.
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
Hi,
Has anybody got the polycom acd function to work? I have the following
setup:
Debian 3.1 - 2.6.8 linux
zlib-1.1.4
libpri-1.2.3
zaptel- 1.2.6
Asterisk - the bweschke/polycom_acd_funtions branch version - I get one
error when doing a make install about needing a newer version of libpri and
On 08:51, Mon 26 Jun 06, Matt wrote:
What on earth is going on with the list?!?! Some of my messages
never make it... then days later I get something like this back:
Final-Recipient: rfc822; asterisk-users@lists.digium.com
Action: failed
Status: 5.0.0
Diagnostic-Code: X-Postfix; mail
On Saturday 24 June 2006 09:44, Paul Hewlett wrote:
I would imagine that this would not solve any problems - the extra overhead
of piping the data over the PCI bus would very quickly negate any speed
gains of the DSP over the native Intel FPU. Additionally you would probably
introduce extra
If I have in my dialplan
[AgentQ]
exten = _XX.,1,Dial(Sip/{$exten},120,g)
exten = _XX.,2,NoOP(here we are)
where [AgentQ] is called by the queue command to a member added by
addqueuemember(Local/[EMAIL PROTECTED])
why don't I get to the NoOp if the agent hangs up during the
announcement
Count me in, my office is in Livonia, but I currently reside in the D. Someone should set up a mailing list for this.--Tom HaydenOn 6/26/06, Michael George
[EMAIL PROTECTED] wrote:Our main office is near Lansing, but we have a person who lives in the
AA are that would like to attend such a
Hello All.
Accurately, my messages also are not being received for the list and the traffic of messages really is very low. It will be a problem of the list? Best Regards
Josué
2006/6/26, Michiel van Baak [EMAIL PROTECTED]:
On 08:51, Mon 26 Jun 06, Matt wrote: What on earth is going on with the
Michiel van Baak wrote:
On 08:51, Mon 26 Jun 06, Matt wrote:
What on earth is going on with the list?!?! Some of my messages
never make it... then days later I get something like this back:
Final-Recipient: rfc822; asterisk-users@lists.digium.com
Action: failed
Status: 5.0.0
Hi Josué
if the Siemens phone calls Asterisk, it didn't get a
dial tone from Asterisk? Is it correct?
if yes, this is depending of Asterisk which didn't
generates a ringback messages as it expexts dial ton
generation localy. So try this workaround for HiPath
local dial ton generation:
- Add
I guess I did not make my point clearly enough. I already do have just
that. An spa-3000 with ALL internal analog phones on it's on FXO. But this
gives just ONE extension for all phones. Yes I could get more FXS's and
run seperate wires.
So with that background what would be nice is a wireless
Yeah, that's what I like about Oz. Everyone knows everyone... miss you guys too!
-Original Message-
From: Rob Thomas [mailto:[EMAIL PROTECTED]
Sent: Monday, June 26, 2006 2:58 AM
To: asterisk-users
Subject: RE: [Asterisk-Users] Asterisk Startups
Well now would be a great time to
Hi Dean -
It should be working. If not, please email me a sip debug trace along
with your /etc/asterisk/agents.conf and your /etc/asterisk/sip.conf.
Thanks.
BJ
On 6/26/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote:
Hi,
Has anybody got the polycom acd function to work? I have the following
'exten-vm'
Jun 26 16:54:02 VERBOSE[8290] logger.c: == Spawn extension (macro-dial, s, 10)
exited non-zero on 'Local/[EMAIL PROTECTED],2'
Jun 26 16:54:02 DEBUG[8290] res_monitor.c: monitor executing ( nice -n 19
soxmix
/var/spool/asterisk/monitor/20060626-165333-1151304813.901-in.gsm
/var/spool
I live in Southfield, our main office is in Pontiac, but our Colo is in
Southfield.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael George
Sent: Monday, June 26, 2006 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL
soxmix
/var/spool/asterisk/monitor/20060626-165333-1151304813.901-in.gsm
/var/spool/asterisk/monitor/20060626-165333-1151304813.901-out.gsm
/var/spool/asterisk/monitor/20060626-165333-1151304813.901.gsm
rm -f
/var/spool/asterisk/monitor/20060626-165333-1151304813.901-*
)
Jun 26 16:54:02 DEBUG
I thought it might be an inadvertent button press, but none of the keys
(on my phone at least) are recognized by Asterisk as fax tones. This
has happened to two different users getting calls from different people
using different equipment. Does anyone else see this behavior
occasionally?
Paul
Hi all.
Today I have tried to connect to the AMP with http://myserverip but I can
not connect to the AMP (it sends me out of my network).
What would be happening?.
The last thing I did is to try to change the digital receptionist manually.
Is there a way to re-install the amp?
Thanks
I'm also in the area, near Southfield. I'd be interested as well.
On 6/22/06, BerkHolz, Steven [EMAIL PROTECTED] wrote:
I am thinking of getting an asterisk user group together for either SEMichigan or just Metro-Detroit.
How much interest in asterisk in Michigan is there on this list?I am already
I am not sure, I will check. If I dont', and get it started, will it just start working? If not, what do I need to do?ThanksJoshua West [EMAIL PROTECTED] wrote: Do you have the Caller ID feature with your telephone service package?sdgesa gaeharth wrote: How can I get the external caller id to
I do get random DTMF tones.
They have been to sparse to diagnose if there was anything common with those
calls.
When it is noticed and I look it up in the logs, it may be any digits.
We see this on zap(PRI) to zap(PRI) bridged calls too.
We are using a TE411P.
--
--
Steven
Paul A. Pringle wrote:
I thought it might be an inadvertent button press, but none of the keys
(on my phone at least) are recognized by Asterisk as fax tones. This
has happened to two different users getting calls from different people
using different equipment. Does anyone else see this
yes. Wind whistling in a car, female voices at a particular pitch and
volume, fax machine running in the background of a voice call with the
speaker on. It happens. Whether this is a problem or not depends on your
pain threshold. I get a couple reports a week, which means that it actually
happens
I also had an intermittent problem, on average one or two faxes a week, that
were not recongnzed as a fax. Then I switched phone companies and have not had
that problem since. It has been over 2 months. I addition, my echo problem
has been practically eliminated and overall voice quality is
Hi to all,
i'm wondering to realize a dynamic macro that can take the number of
extensions to RING,the ring type and all the parameter in a dynamic way.
I have done this code to test it:
macro pbx-ring-group-ael(pbx_id,num_int,ring_type,timeout,ext_string) {
//; pbx_id = Id of PBX in the DB
Hi Richard.
Thank you very much for its attention. In the reality what is occurring is that in some originated calls of the HiPath with destination to the Asterisk they are being without the dumb andrings. I do not have this parameter in my HiPath 4000, what I have seemed in the COT is TR6T (1tr6
Yes, which is why I disable faxdetect entirely. My sister-in-law was
constantly being detected as a fax machine several minutes into
conversations with my wife. As funny as that may seem at first ...
those two eventually make it a not-so-funny situation for me.
lol, Spousal Acceptance
Surely once the call has been bridged the fax detection should turn off ?
Julian
Colin Anderson wrote:
yes. Wind whistling in a car, female voices at a particular pitch and
volume, fax machine running in the background of a voice call with the
speaker on. It happens. Whether this is a problem
Hi,
As you can see in this log the problem is very new to me..
Connected to Asterisk 1.2.5 currently running on volcano (pid = 7874)
volcano*CLI set verbose 4
Verbosity was 0 and is now 4
-- Starting simple switch on 'Zap/2-1'
-- Executing Dial(Zap/2-1, SIP/180|60) in new stack
--
Ok, I count at least 4. Just lets propose when
where for the first meeting group, and start to think about issues
discussion.
Tom, what to we need for the mailing list? I can do
something about that.
Carlos Alperin
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon
Hi,
I'm using the latest 1.2 release of Asterisk and I've noticed that one of the
releases of Zaptel or Asterisk in the past few months seems to have
introduced a problem with MeetMe. Here are the symptoms:
- Very high volume for internal IP phone users
- Very low volume for incoming analog
Jun 26 12:43:16 WARNING[31148]: chan_zap.c:8386 pri_dchannel: Ring requested on unconfigured channel 0/16 span 1Inoticed this message in the CLI, when I tried to effect one call of HiPath 4000 for asterisk. Ring occurred, however when the voicemail of asterisk took care of call it was dumb,
Hey all,
having a terrible time with asterisk-stat -- it runs, server is fine, but
some of the pages don't display properly/at all --- I think this is a code
problem with them, but not sure. I thought everyone loved the asterisk-stat
package?
See below problems. Any ideas? Areski hasn't
Surely once the call has been bridged the fax detection should turn off ?
I'd like to find out a way it can be done, can anyone else comment?
___
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To UNSUBSCRIBE or
26 jun 2006 kl. 10.54 skrev Jean-Michel Hiver:
Johansson Olle E a écrit :
26 jun 2006 kl. 07.10 skrev Martin Joseph:
On Jun 25, 2006, at 4:11 PM, Jean-Michel Hiver wrote:
Hi List,
Is there a way to tell asterisk to only accept SIP streams from
the same IP address that is used for
Has anyone
successfully registered a moto vt1005 to asterisk. If so,
how?
Brandon Warner
Assistant Director of NOC Services
Dark Fiber Solutions
600 1/2 Grant Ave.
York, NE 68467
Office: 402-362-3334
Cell:402-366-2087
"The information
transmitted is intended only for the person or entity
Is anyone getting '500 Internal Server' errors back from their Polycom phones
when Asterisk sends a SIP NOTIFY message to them?
I called Polycom tech support, who where utterly useless.
Of course Polycom won't officially support it anyway, as they only support
Asterisk Business Edition. We're
Hi all,
Could someone point at resources for running Asterisk behind
a firewall.
STUN keeps coming up but, alas, Im easily confused. J
Ray
___
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Asterisk-Users mailing list
To
thanks
we are planing to have around 50-60 users in 1 room.
We've had over 100 participants spread across 30 meetme rooms on a
single server before, and the most we've had in a single meetme room
is 46. I don't know of a hard limit for meetme participants and I
haven't seen a limit in the
On Jun 26, 2006, at 5:51 AM, Matt wrote:
What on earth is going on with the list?!?! Some of my messages
never make it... then days later I get something like this back:
I'm hoping this was a transient issue. I saw this too with a couple of
posts, but it's been ok since.
Marty
Hi, I'm having an issue with a soekris net4801 board and a S101i IAXy
device. When I connect a successfully provisioned IAXy directly via a
crossover cable into an ethernet port of the soekris, the link led turns
on orange so i'ts 10Mb and the activity led blinks like if there is some
action
Hi List,
Just a little question about a notice from asterisk I don't understand:
Here is what I have as soon as I place a call on a E1 line with an a104D
Sangoma Card ( asterisk 1.2.9.1 ) :
Jun 26 18:57:24 NOTICE[16489] channel.c: Don't know what to do with
control frame 15
Does Anyone
On Jun 26, 2006, at 7:14 AM, Doug Crompton wrote:
I guess I did not make my point clearly enough. I already do have just
that. An spa-3000 with ALL internal analog phones on it's on FXO.
That's wrong, phones hook to an FXS.
But this
gives just ONE extension for all phones. Yes I could get
Tzafrir Cohen wrote:
On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:
Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
externo?
É freeware?
Podemos seguir com o projeto Asterisk-PT?
English, please, folks.
I don't know Portuguese
On 6/22/06, BerkHolz, Steven [EMAIL PROTECTED] wrote:
I am thinking of getting an asterisk user group together for either SE
Michigan or just Metro-Detroit.
I'm in Ann Arbor and would be interested in such a group; if you
create a mailing list for it, could you please add me?
Thanks,
Rusty
Douglas Garstang wrote:
Is anyone getting '500 Internal Server' errors back from their Polycom phones
when Asterisk sends a SIP NOTIFY message to them?
Yes, for quite a while. Happens for us, when you do a transfer via the
Polycom's transfer button. Doesn't seem to cause any problems
Yes. It does not seem to cause any problems.
Douglas Garstang wrote:
Is anyone getting '500 Internal Server' errors back from their Polycom phones
when Asterisk sends a SIP NOTIFY message to them?
I called Polycom tech support, who where utterly useless.
Of course Polycom won't officially
On Jun 26, 2006, at 9:32 AM, Raymond Tant wrote:
x-tad-smallerHi all,/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerCould someone point at resources for running Asterisk behind a firewall./x-tad-smallerx-tad-smallerSTUN keeps coming up but, alas, I’m easily confused.
It will be interesting to see how many standards get broken, and how
many proprietary hooks get thrown into the pot. The bean counters smell
some money, and their OS franchise is waning:
http://www.nytimes.com/2006/06/26/technology/26soft.html
--
This message has been scanned for viruses and
I have been seeing the same errors here with Polycom 501 and 601 phones. Asterisk version is 1.2.9.1 and Polycom SIP version 1.6.3On 6/26/06,
Douglas Garstang [EMAIL PROTECTED] wrote:
Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY
I am trying to
register a motorola VT1005. I have many supura ata's that work fine. Anyhelp,
would be great.
Brandon
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do you have the php-gd package installed on your * server?
Chris Earle (CBL) wrote:
Hey all,
having a terrible time with asterisk-stat -- it runs, server is fine, but
some of the pages don't display properly/at all --- I think this is a code
problem with them, but not sure. I thought everyone
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, June 26, 2006 11:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] '500 Internal Server' Error on
SIP NOTIFY
Douglas Garstang wrote:
Is anyone getting
yep
I don't know exactly which things the php-gd is used for, but like I said,
someof the pages work, like the main record page, the little red bars
showing call volume work fine
Really annoying, cause it looks so good at that point, then you go to use
the other pages/features and it's broken
Tzafrir Cohen wrote:
On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:
Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
externo?
É freeware?
Podemos seguir com o projeto Asterisk-PT?
English, please, folks.
Let them talk. What's it hurt
Is there a way to get asterisk to send you a email when it looses or an
extension doesn’t re-register
Roger Workman
Business Development
Upperclassman/Universal Holdings LLC
Voice: 304.324.3800
Fax: 304.324.3801
ICQ: 4447584
Website: http://www.upperclassman.net
Billing Questions: billing at
Sorry to all,
Now only English speaking :)
Your translation was perfect.
Thanks once more
On 6/26/06, Mike Fedyk [EMAIL PROTECTED] wrote:
Tzafrir Cohen wrote:
On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:
Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito
On Mon, 2006-06-26 at 13:16 -0400, Brian Capouch wrote:
It will be interesting to see how many standards get broken, and how
many proprietary hooks get thrown into the pot. The bean counters smell
some money, and their OS franchise is waning:
please type in google.com:
STUN server ALG
The fourth result is a good and small explanation.
On 6/26/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Jun 26, 2006, at 9:32 AM, Raymond Tant wrote:
Hi all,
Could someone point at resources for running Asterisk behind a
firewall.
STUN keeps
Yes we have been getting this error message '500 Internal Server' errors back from their Polycom IP-601 (normally IP address). Do not know why. Was able to regenerate the same issue some times, but not all the time. It is not consistent.
Symptom:
If you have several phones online (10
Daniel Salama wrote:
Dustin,
any updates on this?
Thanks,
Daniel
Hey Daniel!
Yes - just posted the link.
I appologize for the delay.
Here's the link to the forum as well, if anyone is interested. This
should compile and run on Asterisk-1.2.4 and higher.
Sorry to all.
Speaking English only.
Regards
Josué
2006/6/26, Marco Mouta [EMAIL PROTECTED]:
Sorryto all,Now only English speaking :)Your translation was perfect.Thanks once more
On 6/26/06, Mike Fedyk [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué
On Mon, June 26, 2006 20:06, Brian Capouch said:
Tzafrir Cohen wrote:
On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:
Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
externo?
É freeware?
Podemos seguir com o projeto Asterisk-PT?
English,
Beautiful. Will test and give you comments.
Nice work.
- Daniel
On Jun 26, 2006, at 2:55 PM, Dustin Wildes wrote:
Daniel Salama wrote:
Dustin,
any updates on this?
Thanks,
Daniel
Hey Daniel!
Yes - just posted the link.
I appologize for the delay.
Here's the link to the forum as well,
Francesco Peeters (Asterisk) wrote:
Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig
Engels praten!
;-)
Pues my punto fue que un poquito de correo en otro idioma no hace daño,
y si ayuda mucho y molesta poco, ¿por qué quejarse?
B.
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On 6/26/06, Josué Conti [EMAIL PROTECTED] wrote:
OK Marco, irei efetuar os testes.
Se você quiser, posso lhe ajudar no forum, estou a disposição.
Assim que você criar as contas avise para podermos já ir colaborando.
Saudações
JosuéThe differences of licenses are here:
On Mon, June 26, 2006 21:39, Brian Capouch said:
Francesco Peeters (Asterisk) wrote:
Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig
Engels praten!
;-)
Pues my punto fue que un poquito de correo en otro idioma no hace daño,
y si ayuda mucho y molesta poco, ¿por
Hi, guys, I used /usr/src/asterisk/agi/eagi-test.c script to test AGI API, but that script could not print out message tostderr.
any ideas?
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what do you mean by could not print out message to stderr???
Try being more descriptive about your problem. Error messages, how
have you tried etc.
On 6/26/06, Zichao Wu [EMAIL PROTECTED] wrote:
Hi, guys, I used /usr/src/asterisk/agi/eagi-test.c script to test AGI API,
but that script could
Hi All. Please, we need to have more respect with the list. Regards
Josué
2006/6/26, Francesco Peeters [EMAIL PROTECTED]:
On Mon, June 26, 2006 21:39, Brian Capouch said: Francesco Peeters (Asterisk) wrote:
Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten!;-)
All of the Asterisk Say applications have stopped working.
Example: SayDigits(), SayNumber(), etc...
CLI output:
-- Executing SayDigits(SIP/209.247.17.5-b7901508, 12356) in new
stack
== Spawn extension (facloc-english, 12356, 2) exited non-zero on
'SIP/209.247.17.5-b7901508'
This is
-Original Message-
From: Moises Silva [mailto:[EMAIL PROTECTED]
Sent: Monday, June 26, 2006 2:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AGI script can not print out
error message
toconsole
what do you mean by could not
Check /var/log/http/error.log
Usually, asterisk-stat fails because it tries to use more memory than
allowed in php.ini.
Julian J. M.
On 6/26/06, Chris Earle (CBL) [EMAIL PROTECTED] wrote:
yep
I don't know exactly which things the php-gd is used for, but like I said,
someof the pages work,
Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig
Engels praten!
;-)
Pues my punto fue que un poquito de correo en otro idioma no hace
daño, y si ayuda mucho y molesta poco, ¿por qué quejarse?
Quel bordel, sacrebleu!
--
Jean-Michel Hiver - http://ykoz.net/
Découvrez
Carlos Alperin wrote:
I live in Southfield, our main office is in Pontiac, but our Colo is in
Southfield.
I'm here in Sterling Heights, have a call center in Clinton Twp that's
100% Cisco/Linksys phones (7940s and SPA942s) and rent in a colo down in
Southfield as well where I connect to
Nicolás Gudiño wrote:
Hi Ronald,
If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are many
other cases.
You should install php-pcntl (or compile php to add support for
process control functions). The inuse
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