Hey Dan, and thanks a lot for your answer regarding Cisco CCM and MTP. I
will continue my tests, and maybe give a try to the patch you
mentionned. However, this will probably be too cutting edge for the
project ;-) I have a few questions, though:
- You mention that Cisco indicates that any H323
On Monday 31 October 2005 20:14, Dean Collins wrote:
Is this for real? Can echo be codec induced or are they spinning me a
line?
Kris has given the most reasonable answer but I'd like to throw in the
possibility that poor codec implementation may cause additional delay which
is pushing the
In sip.conf I have this:
[1000]
type=peer
host=dynamic
defaultip=192.168.62.100
dtmfmode=rfc2833
mailbox=
context=dialoutcont
callerid=Folkert van Heusden [EMAIL PROTECTED]
You've defined a peer - which is for calls TO that endpoint. You need a
user if you want to accept
Password = does not exist.
The reason it is probably working is that by using that you are specifying no
password for the account. Check your settings.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
Hi,
When we use mISDN and Asterisk some times after Dial the PSTN
doesn't send Ringing Media. The indication is coming and is displayed
on Asterisk CLI but there is not sound. This is very unpleasant for
the user which can't see the Asterisk CLI to know that the call is in
Rining State.
Hi,
How can be done similar functionality like Panasonic PBX and Phones
where for incoming calls from PSTN the ringing type is different with
this that is from some internal extension. For example we can
configure Asterisk PBX to add some special character before or after
the number in
Hi,
I would like to program some PSTN lines to be indicated on the
phone (GXP-2000) when are busy and etc. How can we do that using
Speed Dial/Configurable line indicators or/and Line 1-4 Keys?
How Asterisk will inform the phone when the line is busy or
ringing, etc.?
Best Regards,
IAX2 trunking don't work with slinear codec. It's sounds like byteswapping
bug. When I change codec to alaw it works fine. When I disable trunking
it also works.
first server -- Opteron (32 bit mode)
second server -- Athlon
At first server it reproduced, e.g. with Playsound. At second server I
We did a bunch of audio quality tests with slinear some weeks ago, it
seems completely broken. (not limited to iax2 trunking)
It gave us extremely bad results in 30% of all calls, iax2, sip, jb, no
jitter buffer.
Its on my todo list to find out why this happened and file a bug report
if
Hi
That seems to work fine now using Z for the last line. Thanks very much for
your help and explanations.
Incidentally, we are using VoipTalk but are looking to trial another
provider as have been experiencing the occasional call cut out and quality
issues. Out of interest, do you have
Hello list,
I don't know if this is the right place for this question, but I've seen that a few of you have been using Polycom IP phones...
My problem is that I can not get any XHTML page to show on the
browser... I've done a simple page that only displays testpage, and
tested it with w3.org's
Steve Davies wrote:
On 31 Oct 2005, at 08:25, yusuf wrote:
Hi all,
I currently have this configuration.
exten = _X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5060)
exten = _X.,102,Set(PRI_CAUSE=42)
exten = _X.,103,Hangup()
I have an Asterisk box connected via E1/PRI to Siemens PBX. The
I just went thru the docs and UPGRADE file.
Is there any other place that has a more detailed description of the
changes/additions on this new version? For example, new apps, changes in the
dialplan, New dialplan functions replacing old applications?
|-Original Message-
|From: [EMAIL
Anybody already tried compiling spandsp with the new 1.2beta2?
How about unicall?
___
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Tuesday 01 November 2005 08:43, Kevin P. Fleming wrote:
The second beta of Asterisk 1.2.0 has been released! It is available from
the ftp.digium.com FTP servers, as well as the Digium CVS servers (under
the 'v1-2-0-beta2' tag).
I think you forgot to place it to ftp server.
We're having a meeting at 10am GMT in the English conference rooms (691) to
discuss the new beta (installation problems/requests etc).
The conference rooms (IAX2) are available at http://freevoip.gedameurope.com
If anyone would like a free hosted conference room for an Asterisk related
meeting,
Extremely basic Hello World type page.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
Title: NetConcepts Polycom home page
Google
Yahoo WAP
Amazon WAP
Hello asterisk users,
I want to register sip agents (polycom ip 300) and
asterisk on Ser (sip express router)
sip user1--SERsip user2
|
|
Asterisk
How may i configure Ser+Asterisk in order to provide
Moh to sip agents
Do you know if polycom's 301 or 501 (which I think don't have minibrowser)
can support pushing xml info pages to the display so it can show info like
weather, etc?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Chris Mason (Lists)
|Sent: Tuesday,
Freely available for download at:
http://www.asteriskguru.com/tools/queue_stats.php
Changes since last version :
- Bugs removed (and new bugs introduced)
- pdf reporting.
- internationalisation (language files)
- works on both windows and linux
Currently only few language files are included,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
my topology:
CME (cisco callmanager express 12.3(14)T4 on 1751v): 192.168.17.1
Asterisk (1.0.9 on Freebsd 5.4): 192.168.17.10
from 12.3(11)T6 and later CME sends num$ and not num only on sip
trunks. See the 'sh sip reg status':
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
mmm
a solution is maybe something like that?
http://mail.iptel.org/pipermail/serusers/2005-May/019677.html
Regards
Andrea
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (Darwin)
Hi all,
For those of you that are interested, the Asterisk patch has been updated to
work with v1.2.0-beta2. It is available separately at :
http://www.lusyn.com/asterisk/asterisk_ukcid_beta.diff
Usage instructions are at :
http://www.lusyn.com/asterisk/patches.html#beta
Rgds,
Marc
Eur
Walter:
Lo que
te pregunta Carlos, (Si no me equivoco), es de que parte de EEUU quieres que
sean los números "Entrantes" (Para que puedan llamarte a ti)
Saludos.
Juan.
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]En nombre de Walter
WillisEnviado el: Lunes,
Hello,
Is this possible to send SIP messages (MESSAGE,
SUBSCRIBE, NOTIFY) to a sip proxy from asterisk ?
Regards
Harry
___
Appel audio GRATUIT partout dans le
This is asterisk 1.2 beta.
It seems a certain voicemail box wouldn't accept any more messages when
there were 100 messages in it...but I can't find where this is
configured. Is this a hard coded limit or can it be changed in
voicemail.conf?
Also, does asterisk automatically delete
maxmessage= in voicemail.conf Asterisk does not automatically delete voicemails after a period of time.On 11/1/05, Adam Moffett
[EMAIL PROTECTED] wrote:This is asterisk 1.2 beta.It seems a certain voicemail box wouldn't accept any more messages when
there were 100 messages in it...but I can't
So maxmessage is a global setting for all voicemail boxes?
BJ Weschke wrote:
maxmessage= in voicemail.conf
Asterisk does not automatically delete voicemails after a period of time.
On 11/1/05, *Adam Moffett* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
This is asterisk 1.2 beta.
John Novack wrote:
Rich Adamson wrote:
Assuming you are using the TDM card, there is no code in asterisk to
detect whether a pstn line is connected/disconnected, nor does it
listen for dialtone before dialing.
And for some reason this isn't considered a SEVERE defect?
If the battery
Hi all..
I just setup a test box with Debian running kernel 2.6.
Went to CVS and did a checkout of the new beta 2 release
using the command: cvs checkout -r v1-2-0-beta2 zaptel
libpri asterisk asterisk-addons asterisk-sounds.
I then compiled libpri fine and moved on to zaptel. Did a
make
I agree, I would definitely love to find out more about a lot of the
features, new apps (MixMonitor?), etc... I did full text searches against
the tree and couldn't find a single reference to mixmonitor...
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On
Hi,
Can I define user permission for show command only, I only want user to
check how is registered right now.
Best Regards,
Ishay.
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Asterisk-Users mailing list
i heard some talk about something in zaptel is currently
incompatible
with 2.6.13.
is this so?
if so, will this be fixed soon?
zaptel 1.0.9.2's release notes has something about 2.6.13 kernels.
I was thinking more about the CVS HEAD version
My machines are totally CVS HEAD and have been
Walter,
No se acerca de que es lo
mas atractivo. El servicio puede ser en el horario que tu quieras (Nosotros
trabajamos 24 hs, 7 dias a la semana, 365 dias al año), pero si quieres restricción
de horario, se puede hacer.
No dije nada acerca del
español
Carlos
From:
What about vi app_voicemail.c?
#define MAXMSG 100
#define MAXMSGLIMIT
Thanks,
Steve
So maxmessage is a global setting for all voicemail boxes?
BJ Weschke wrote:
maxmessage= in voicemail.conf
Asterisk does not automatically delete voicemails after a period of
time.
On
Walter,
Quiero saber si tenes preferencia
por algun codigo de area especifico, o cualquier codigo de area te da igual.
Las llamadas van a ser
solo salientes, o pensas recibir llamadas entrantes tambien?
Saludos,
Carlos Alperin
Senior System Engineer
Seneca Communications,
On Tue, 2005-11-01 at 14:54 +0100, Roy Sigurd Karlsbakk wrote:
i heard some talk about something in zaptel is currently
incompatible
with 2.6.13.
is this so?
if so, will this be fixed soon?
zaptel 1.0.9.2's release notes has something about 2.6.13 kernels.
I was thinking more
AFAIK, the official language of this mailing list is English.
On Tue, 2005-11-01 at 08:54 -0500, Carlos Alperin wrote:
Walter,
No se acerca de que es lo mas atractivo. El servicio puede ser en el horario que tu quieras (Nosotros trabajamos 24 hs, 7 dias a la semana,
Try this:
phoenix*CLI show application mixmonitor
phoenix*CLI
-= Info about application 'MixMonitor' =-
[Synopsis]
Record a call and mix the audio during the recording
[Description]
MixMonitor(file.ext[|options[|command]])
Records the audio on the current channel to the specified file.
If
I don't use quotes on either if that makes a difference. When are you
setting it? Maybe you are losing the incoming number by the time you
set it the number has been changed to the extension. I use Sipuras and
have no problem with this. Here's an example of a macro I use when
forwarding an
Hi,
I want to transfer a call that has come into one queue, and that I have
already accepted, into another queue.
When I try this asterisk tells me Transfer attempted with no
appropriate bridged calls to transfer.
It is possible to forward the call to another person, but forwarding
into a queue
Hi all i have a question. is my first time using [EMAIL PROTECTED] and i
need your help
i configure all my asterisk to go outside and work perfect via te110p
but now i need to receive calls. but when in my PBX i digit the number
for example 202 the asterisk receive a s i suppouse. the error
I'm sorry. Hadn't reached my second cup of coffee before I wrote that first message. :-( Correct. maxmessage isn't currently set in the code to be read anywhere other than [general]. HOWEVER, maxmessage= is for the maximum length of a message, not how many individual messages.
You're looking for
Hello,
I read roadmap on www.openpbx.org.
Does chan_exosip2 will be able to provide a real sip
proxy ?
What about asterisk solutions ?
Harry
___
Appel audio GRATUIT partout dans le
On Tuesday 01 November 2005 12:19, Anton Krall wrote:
Anybody already tried compiling spandsp with the new 1.2beta2?
How about unicall?
I did, and it works fine.
What is unicall?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Sergey Okhapkin wrote:
AFAIK, the official language of this mailing list is English.
Butt out. What's the difference to you if two others want to talk in
their native language?
English is for sure the language that most of us on the list would
prefer to use, but Asterisk is a world-wide
On Tuesday 01 November 2005 15:45, Robert Webb wrote:
Hi all..
I just setup a test box with Debian running kernel 2.6.
Went to CVS and did a checkout of the new beta 2 release
using the command: cvs checkout -r v1-2-0-beta2 zaptel
libpri asterisk asterisk-addons asterisk-sounds.
I then
the i is if you were to press an incorrect digit. s is for START. You
can also specify your DID as a start point.
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-Original Message-
Sir,
If the people had asked questions on
Spanish, I dont have any problem on answer those questions.
Not everybody speaks English, and I didnt
know any rule that forbids any other language.
Sorry,
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergey
Anyone have any hints on how to get the polycom ip501 to send dtmf
inband, our upstream providers require inband and the native rfc2833
format of the polycom does not work.
Is there a way to get asterisk to send inband to a SIP peer when it
receives rfc2833 form a sip user?
Thx!
Hi All -
I'm calling people on Zap interface using /var/spool/asterisk/outgoing
and then putting them into a MeetMe. This works 100%, but tends to
give unknown name and number on the meetme list command...
eg:
User #: 01unknown no nameChannel: Zap/1-1
(unmonitored)
I really
In article [EMAIL PROTECTED],
Robert Webb [EMAIL PROTECTED] wrote:
Hi all..
I just setup a test box with Debian running kernel 2.6.
Went to CVS and did a checkout of the new beta 2 release
using the command: cvs checkout -r v1-2-0-beta2 zaptel
libpri asterisk asterisk-addons
Hello Harry,
This is rather the wrong list to ask this... since this is Asterisk, not
OpenPBX.org
Chan_exosip2 though is something I'm basically designing to have 3 operating
modes.
Full server: Most closely resembles chan_sip in that it acts as a B2BUA
Partial proxy: Extensions are mapped to
Si señor, I AGREE.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch
Sent: Tuesday, November 01, 2005 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana
You will probably also need to change the media exchange timers in CCM
if you are going to use it as a PRI gateway - otherwise asterisk - 323
- CCM - PSTN calls will get dropped after 4 secs of ringing.
On Mon, 2005-10-31 at 14:41 +0100, Patrick Zwahlen wrote:
Hey Dan, and thanks a lot for your
I'm trying to create a call file for Asterisk, but keep getting an
error and I can't figure out why. Any thoughts?
DEBUG FROM ASTERISK:
Nov 1 10:06:28 WARNING[24104] pbx_spool.c: Unknown keyword ' Channel'
at line 5 of /var/spool/asterisk/outgoing/callback
Nov 1 10:06:28 WARNING[24104]
Damon Estep wrote:
Anyone have any hints on how to get the polycom ip501 to send dtmf
inband, our upstream providers require inband and the native rfc2833
format of the polycom does not work.
Is there a way to get asterisk to send inband to a SIP peer when it
receives rfc2833 form a sip user?
In article [EMAIL PROTECTED],
Tony Mountifield [EMAIL PROTECTED] wrote:
No, the problem is that you don't have the kernel development files
installed. I don't know what you need to apt in Debian: in Fedora
you would need to install kernel-devel or kernel-sources depending
on which release of
While I don't disagree in principle, I think an issue is that much of the
benefit of this list is the knowledge gained by reading about other people's
problems and resolutions. If these discussions start being held in other
languages we will not all be able to benefit from them.
JMO
Bill Hunt
Yo tambien.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 1 Nov 2005, Carlos Alperin wrote:
Si se?or, I AGREE.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch
Sent: Tuesday, November 01, 2005
I then compiled libpri fine and moved on to zaptel. Did a
make clean then make install and get the following error:
How about `make linux26' ?
I am not up to speed on make or its errors, but it looks
like to me that it is complaining about /usr/src/zaptel
not
Thanks for reply,
Does Chan_exosip2 is stable where may i find help ?
Chan_sip from asterisk doesn't support IM and presence
via SIMPLE.
I whish to use either asterisk or openpbx to provide
telephony features with SER to relay IM and presence
SIMPLE
openpbx or asterisk
On Tue, Nov 01, 2005 at 02:46:06PM +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Robert Webb [EMAIL PROTECTED] wrote:
Hi all..
I just setup a test box with Debian running kernel 2.6.
Went to CVS and did a checkout of the new beta 2 release
using the command: cvs
In article [EMAIL PROTECTED],
Matt [EMAIL PROTECTED] wrote:
I'm trying to create a call file for Asterisk, but keep getting an
error and I can't figure out why. Any thoughts?
DEBUG FROM ASTERISK:
Nov 1 10:06:28 WARNING[24104] pbx_spool.c: Unknown keyword ' Channel'
at line 5 of
http://lists.digium.com/mailman/create
This list supports English (USA). Possibly
our spanish speaking friends need their own list?
Thanks,
Steve
- Original Message -
From:
Carlos
Alperin
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I ran into a problem with 1.2-beta2 with using realtime sip. It looks
like chan_sip.c has a new field fullcontact. I just added this column
to my database as a VARCHAR(128) and things seem to be working now.
I just thought someone else might want to
Mark:
Thank you ... I will try this.. However, before I do.. You say you call
this BEFORE you dial the forward.. do you mean just before the regular
DIAL command... What is your DIAL command for your Sipura if I may ask.
Thanks.
Ben.
I don't use quotes on either if that makes a difference.
Is there a way to get a thread ID (???) in the log file?
I see the process ID, but I think some way to correlate which items are tied
to which items would be helpful for troubleshooting.
When there are multiple simultaneous calls going on, It takes a lot more
effort to correlate hang-ups and
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Tuesday, November 01, 2005 8:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] inband dtmf on ploycom ip501?
las llamadas serian solo salientes, por lo que se, pero que quieran el
servicio de recepcion de llamadas, cuanto costaria todo eso???
podrias darme un costo de loq ue ofreces?El día 1/11/05, Carlos Alperin [EMAIL PROTECTED] escribió:
Walter,
Quiero saber si tenes preferencia
On Tue, 01 Nov 2005 10:18:45 -0500
Paul Zimm [EMAIL PROTECTED] wrote:
I then compiled libpri fine and moved on to
zaptel. Did a make clean then make install and get the
following error:How about `make linux26' ?
I am not up to speed on make or its
senecacom.net es tu dominio??
por cierto no das planes de reseller???
jejejeje.
mi ingles es
El día 1/11/05, Walter Willis [EMAIL PROTECTED] escribió:
las llamadas serian solo salientes, por lo que se, pero que quieran el
servicio de recepcion de llamadas, cuanto costaria todo eso???
podrias
Unicall is Steves Underwood E1 R2MFC support using spandsp.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Roman
|Sent: Tuesday, November 01, 2005 8:23 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users]
Hi list,
at first time, sorry for my english.
For Asterisk community, and especially for spanish speaking, exist
www.asterisk-es.org. I think they are others sites for spanish speaking,
for example in Argentina.
This community have their own list at
http://groups.google.com/group/asterisk-es.
Any special considerations? How did you patch the files? Manually?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Roman
|Sent: Tuesday, November 01, 2005 8:23 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re:
thanks!!!
ask.
the fullcontact varchar(128) is added to
#
# Table structure for table `sip_buddies`
#
CREATE TABLE `sip_buddies` (
`id`int(11)NOTNULLauto_increment,
`name`varchar(80)NOTNULLdefault'',
`accountcode`varchar(20)defaultNULL,
`amaflags`varchar(7)defaultNULL,
Hello
as from 1.2beta2
it's not possible to build chan_capi
we get compile
errors
use_ast_mutex_init_instead_of_pthread_mutex_init
if someone as any
idea to correct this in the source code of chan_capi
bets
regards
Thierry
[EMAIL PROTECTED]
Tel : +33 (0)3 90 40 06 75
Fax: +33 (0)3
Nothing special... yes I patched them manually -- current spandsp patch is
broken a little.
On Tuesday 01 November 2005 18:11, Anton Krall wrote:
Any special considerations? How did you patch the files? Manually?
|
|On Tuesday 01 November 2005 12:19, Anton Krall wrote:
| Anybody already
On Tuesday 01 November 2005 18:11, Anton Krall wrote:
Unicall is Steves Underwood E1 R2MFC support using spandsp.
Well, don't know about that...
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Asterisk-Users mailing list
Can I get a copy of your makefile? I am having a devil of a time
getting it to work..
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-Original Message-
From: [EMAIL PROTECTED]
Thank you,
I hadn't gotten around to actually swapping my prexisting 1.2beta1 system
with the new 1.2.0beta.
Cheers
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Rich Adamson
-Sent: Tuesday, November 01, 2005 9:04 AM
-To: Asterisk Users
I want no echo can on Zap to Zap bridge, but I do want on Zap to Sip bridge.
Zap to Zap is called Native bridge.
Zap to Sip, I never see a bridge created, but I do get Bridge stops
bridging channels SIP/5665-1b6f and Zap/25-1 on hangup.
I am trying this on both Zap spans:
echocancel=no
Roman wrote:
I think you forgot to place it to ftp server.
http://ftp.digium.com/pub/asterisk/asterisk-1.2.0-beta2.tar.gz
gives NOT FOUND
This is being fixed right now... one of the FTP servers was out of sync.
___
--Bandwidth and Colocation
I read roadmap on www.openpbx.org.
Does chan_exosip2 will be able to provide a real sip
proxy ?
What about asterisk solutions ?
I guess you can use chan_exosip2 with asterisk if you hack it in
yourself. Also, as soon as asterisk is released in one single GPL
license, it may as well be
Steven wrote:
Please advise if I can get echo can off with Zap to Zap bridge, but have
echo can on with Zap to Sip bridge.
You have the configuration options specified exactly backwards :-)
___
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I need to get the user portion of the Referred-By URI for a call that
comes
from an SER SIP proxy into Asterisk. I tried using BLINDTRANSFER but
that didn't contain the correct value. Anyone have any thoughts?
_Steve
--
___
--Bandwidth and
Hi,
I currently have several PAP2-NA units configured to an Asterisk
box, everything works fine except from the fact that after dialing a
number I can hear ringing tones. When I connect to the same Asterisk box
using XLite or EyeBeam I hear only one, any ideas on what may be wrong
on the
In fact I want to forward SIP MESSAGES to any sip
proxy
Unless chan_exosip2 is able to relay IM presence via
SIMPLE .
Harry
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a écrit :
I read roadmap on www.openpbx.org.
Does chan_exosip2 will be able to provide a real
sip
proxy ?
What about
Just started getting this warning message about every minute.
ast_sched_runq ran 20 scheduled tasks all at once
I know it's a warning but Mark/Kevin Co must have thought it worth
mentioning.
--
Dave Cotton [EMAIL PROTECTED]
___
--Bandwidth and
Hi all.
We are having some strange behaviors in our asterisk box. A Dual p4
Xeon 3ghz, with 2gb of ram and a a digium TE410p card.
The system is an Asterisk V 1.0.9 with Unicall 0.0.2c (Argentina
variant) to interconnect with an Alcatel pbx and to the pstn.
It has an average of 10 concurrent
The full message not the Google search part.
Just started getting this warning message about every minute.
WARNING[8642]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran
20 scheduled tasks all at once
I know it's a warning but Mark/Kevin Co must have thought it worth
I've been playing with ael a little bit and wondering how much time to
devote to it. Originally, when I read about it, I thought I'd try to
convert my extensions.conf files to ael. But after playing a little
bit, I'm not sure that's a good idea.
A couple of observations and questions:
1. As
Guys.
Can somebody explain a bit further the use of this new feature in
features.conf
[applicationmap]
;testfeature = #9,callee,Playback,tt-monkeys ;Play tt-monkes to
;callee if #9 was pressed
I cant find more info anywhere and I suspect this is
I'm STILL waiting on an answer to my exact same question. That was around 3
weeks ago
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-John Biundo
-Sent: Tuesday, November 01, 2005 1:13 PM
-To: asterisk-users@lists.digium.com
-Subject:
John Biundo wrote:
A couple of observations and questions:
1. As far as I can tell, there's limited interoperability between
existing (extensions.conf) and new (extensions.ael) constructs. For
example, and please correct me if I'm wrong, but I don't believe there's
a way to access an ael
Hello,
Ever since I started playing with Beta versions of Asterisk, I've had a
problem. It might just be coincidence though, since before that I didn't
touch the Asterisk PC for a good 2 weeks and I had done alot playing around
with config files.
I have a 4 port FXS/FXO card (with 2 of
No. in any place, even i haven't any agi script in the dialplan.
exten = _87XX,1,Dial(SIP/${EXTEN},60)
This is the only i have to call the sip ata.
Also, something i forgot to mention, i have in the sip.conf and the
audiocodes the srvlookup disabled. Browsing the problem in the list i
see that
Hello All,
PRI to SIP calls are being destroyed after a few minutes, and I get the
stream below in a full debugging log. SIP to SIP works okay.
I'm using Asterisk CVS head 10-31-2005 with a TE110XP card set to T1.
Actually two TE100XPs are installed, but I only have one T1 plugged in
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
this is from a sip debug on IOS:
Nov 1 19:42:12 192.168.17.1 743:
Nov 1 19:42:12 192.168.17.1 744:
Nov 1 19:42:21 192.168.17.1 745: Nov 1 19:41:29.895 MET:
//-1//SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI
with IP addr:
Hi,
I am trying to figure out how to setup asterisk with a
TDM400 (TDM04B), so that the first 3 lines incoming will be answered and
the 4th line is just for outgoing calls but doesnt answer on incoming
calls.
Anyone know how to set that up?
My zaptel.conf :
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