Nope Tim,
had tried that already, duznt work.
Here's the cli output
===
Executing Set(SIP/4000-097afc90, fwdTime=*|mon-tue|*|*) in new stack
Sep 7 12:22:20 WARNING[24964]: pbx.c:6062 pbx_builtin_setvar: Ignoring
entry 'mon-tue' with no = (and not last 'options' entry)
Sep 7
I have the same problem on on of our systems, but i always thought it to
be a problem in the ATA's connected to this server.
(My customer has a lot of traffic on the lines and only sometimes hears
this problem).
It seemed to happen especially with loud woman voices, but i was unable
to
I see in CLI:
ast_parse_allow_disallow: Cannot allow unknown format 'h264'
What can I do ?
I see on Asterisk home page, that h264 is not listed.
When does Asterisk need h264 at all? If one phone calls another phone,
than it is only passed through and does not need it, or am I wrong here?
Hi friends,Thank you to all for your response and
cooperation to me. I have a doubt.What i do for sending and receiving
Faxusing a fax machine with numberextension = 433 in my
office?
Wich filesto be configured for this
application?
Bye,Andrea
Hi,
I have a Asterisk configuration as follows
SIP(LAN) IAX2(WAN)
PSTN GW *-client -- *-Server
The *-Server serves recorded prompts as part of an IVR menu to the *-Client
I am using the new JitterBuffer in the
Hello,
I look at the new sip firmware however i don't
undanstand the presence features.
I don't use LCS but SER as presence server this one is
able to provide a ressource list server and xcap
server
for sip buddies lists .
Does polycom phones can suscribe to a
sip:[EMAIL PROTECTED] for example
I dont know if Im mistaken or not but I
noticed in a iax2 show peers command that it is showing my iax2 connections as
netmask 255.255.255.255
All of my lan traffic is supposed to be running on
255.255.255.0
Is there a way to change this?
(the reason for asking is the faktortel
Hi,
Nobody has a hint for this?
this seems to be a big problem when
calling!
regards rene
Von: René Enskat [Teamware GmbH]
[mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 6. September 2006
11:39An: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Betreff: mobile refusing call
Hi
Hi Dean
Dean Collins schrieb:
I don’t know if I’m mistaken or not but I noticed in a iax2 show peers
command that it is showing my iax2 connections as netmask 255.255.255.255
/32 are hosts addresses...which is correct.
All of my lan traffic is supposed to be running on 255.255.255.0
This
Hi,
has anybody had success compiling bristuff with kernel 2.6.17.11? Error
messages are below...
Cheers,
Arik
---
/usr/src/asterisk_1.0.10/bristuff-0.2.0-RC8s/zaptel-1.0.10/zaptel.c:6520:
warning: passing argument 4 of 'class_device_create' from incompatible
pointer type
Ok, cool, thought it was probably always that, just having problem with
faktortel at the moment so must be another problem.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Richard Klingler
Sent: Thursday, 7
Hey all,
A previous annoyance with not being able to call out to my brother on FWD
from my Asterisk system had me thinking that since I have my own PBX, and
that system has it's own 1-to-1 static NAT to the internet, I should be
able to act as the provider for him or any of my family, and
According to this thread
http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=990forum=3
Cisco 7970 (SIP 8.0.2) sends wrong request to http server and that is why Cisco
7970 IP Phone doesn't show phone directory or services. It seams there is the
same problem with SIP 8.0.3 firmware.
Tomislav Parčina schrieb:
According to this thread
http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=990forum=3
Cisco 7970 (SIP 8.0.2) sends wrong request to http server and that is why Cisco
7970 IP Phone doesn't show phone directory or services. It seams there is the
same problem
Nick,
Anything helpful in the asterisk or system logs.
Try bumping up the debug and verbose levels see what shows up on the
console.
Weird that it would work inbound and not outbound.
Bob...
On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote:
Hey all,
A previous annoyance with not
You say its not your code. But yet, why would you
actually admit to one of your own leaking it. Well
some research has been done one the code.. here's what
we found..
the g723.1 library code that was posted matches the
library code distributed by Digium and committed to
CVS by Mark in March
You say its not your code. But yet, why would you
actually admit to one of your own leaking it. Well
some research has been done one the code.. here's what
we found..
the g723.1 library code that was posted matches the
library code distributed by Digium and committed to
CVS by Mark in March
Bob,
I will up the logs today, have my phone at work with me. (though the Wife
and Kids are not up yet ;)
Anything specific I should target?
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Thu, 7 Sep 2006, Bob Chiodini
Hi,I have registered with IPKall ang got the number i.e., 206XXX. When I call to this number, It is telling that "The party you are calling is currently busy". Here I am giving my config details.When I registered with IPKall, I entered these below values:SIP Phone number: 7312567SIP Proxy:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Actual problem was with the Phonelabel string being too long (o;
Found out with in the logs...
I'm glad you solved it.
So I'm staying with SIP 8.0.2 as it also supports XML push whereas
the SCCP images don't support it at all...
Yes,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Joe Shmoe wrote:
You say its not your code. But yet, why would you
actually admit to one of your own leaking it. Well
some research has been done one the code.. here's what
we found..
the g723.1 library code that was posted matches the
Nick,I have done what you are talking about as far as being a provider for family members. I used an IAX softphone mainly to eliminate the need for so many holes in the firewall. And secondly because the idefisk IAX softphone allowed me to extract the zip version, configure the phone, and zip the
Crazy Boy wrote:
I have given my total configuration. Please tell me the solution.
Looking forward to your response. Thank you.
You need to also include the output from the console.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Joe Shmoe wrote:
You say its not your code. But yet, why would you
actually admit to one of your own leaking it. Well
some research has been done one the code.. here's what
we found..
When and where did KPF admit to it being Digium's code?
-
Von: Crazy Boy [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 7. September 2006 14:25
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Incoming call problem-calling part is busy(IPKall)
Hi,
I have registered with IPKall ang got the number i.e., 206XXX. When I
call to this
Hello, All.
I've been lurking on this list for some time, trying to drink from the
fire hose. Now, I have a few questions. First, though, here is the
background:
I work for a testing facility where we test telephony products. We have
been using Asterisk for about 4 months now as a test bed
Yes, it seems to be happening on any call that passes over the T1 card.
SIP-to-SIP works fine.
Date: Thu, 07 Sep 2006 10:36:24 +0300
From: Zoa [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Volume events causing talk off on
Asterisk with Digium 411P
To: Asterisk Users Mailing List -
On Thursday 07 September 2006 08:45, Matt Riddell (IT) wrote:
When and where did KPF admit to it being Digium's code?
Via psychic vibrations, obviously.
-A.
___
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asterisk-users mailing list
To
Bruce,
I *just* tested the XtremePhone, IAX2 softphone. Other than trying to
figure out how to get it to send proper CallerID to the other phones, it
worked right off, in both directions. Excellent!
Perhaps working the IAX2 angle will be less of a hassle, I will go looking
for one that
You say its not your code. But yet, why would you
actually admit to one of your own leaking it. Well
some research has been done one the code.. here's what
we found..
the g723.1 library code that was posted matches the
library code distributed by Digium and committed to
CVS by Mark in March
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
My 7970G running 8.0.2 SIP firmware works perfectly with
the Open XML 79xx directory frontend...
I have never tried Open XML 79xx, although I have hear of him.
Also can can push XML alarm messages to the phone
from nagios system.
Can
Tomislav Parčina schrieb:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
My 7970G running 8.0.2 SIP firmware works perfectly with
the Open XML 79xx directory frontend...
I have never tried Open XML 79xx, although I have hear of him.
http://www.asteriskpbx.de/index.php?open79xx
Can we apply netmask on SIP Context instead of individual IP address?Thanks,Dean Collins [EMAIL PROTECTED] wrote: Ok, cool, thought it was probably always that, just having problem withfaktortel at the moment so must be another problem. Cheers,Dean -Original Message- From: [EMAIL
What were the proposed changes to VPM_DEFAULT_DTMFTHRESHOLD
?
-- -- Steven
http://www.glimasoutheast.org
"Servetas, Andrew" [EMAIL PROTECTED]
wrote in message news:[EMAIL PROTECTED]...
We are experiencing random talk
off events when we hear a loud volume event on the PSTN
On Thu, 2006-09-07 at 02:31 -0700, Joe Shmoe wrote:
You say its not your code. But yet, why would you
actually admit to one of your own leaking it. Well
some research has been done one the code.. here's what
we found..
the g723.1 library code that was posted matches the
library code
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
http://www.asteriskpbx.de/index.php?open79xx
http://www.voip-info.org/wiki/index.php?page=Cisco+79XX+XML+Push
I'll have to check on those two.
Would be good to know what the actual text output is to
compare with mine...
Discovered
Does anyone know how many active channels can support for transcoding ulaw to G729 by using 4x 3.6GHz Xeon Processors?Thanks,
Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.
___
--Bandwidth and Colocation
Hi,This is a sample file I am currently using on my server. My server has a public IP address and an internal IP address (duan NIC). It runs Fedora Core 3 running iptables firewall already configured with ports 4569, 5060, 1-2 open (udp and tcp)
Polycom phones send a SIP SUBSCRIBE message for buddy watching.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 07, 2006 4:15 AM
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: [asterisk-users] New polycom firmware /
Andrew Kohlsmith wrote:
On Thursday 07 September 2006 08:45, Matt Riddell (IT) wrote:
When and where did KPF admit to it being Digium's code?
Via psychic vibrations, obviously.
It's not Digium's code, IIRC. It's ITU code. You can download the ITU
reference code (in C) from the ITU for
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Kokfoo Soo wrote:
Does anyone know how many active channels can support for transcoding ulaw to
G729 by using 4x 3.6GHz Xeon Processors?
In one machine?
I'd guess at around 200-300 absolute max if the calls are spread evenly
across CPUs.
Normal
Alberto Sagredo wrote:
I use canreinvite=yes in my config files, and it does work, so maybe its
a spa 941 misconfiguration.
I think if nat=no sometime it has problems if you are behind NAT, but
under same network it must not fail.
I am behind a NAT, though the whole network is seperate,
My last update was a while back and as I remember svn trunk did not
compile and I was advised to use branches 1.2 till further notice.
Have I missed the further notice and can we use now svn trunk or is the
advice still to use branches 1.2 ???
bye
Ronald
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Is there a way to have asterisk failover to another codec when you're
out of g729 licenses? I did some google searching and all I could find
was this post from early 2005.
http://lists.digium.com/pipermail/asterisk-dev/2005-February/009405.html
Has
I'm wondering if this is a bug in voicemail...
User A has elected to receive email notifications of voicemail and also have
the original voicemail deleted from the server, such that the WMI light is
never lit. If user B forwards a voicemail to user A (via the option in
voicemail), then user A
The directed call pickup functionality is
turned off by default you have to explicitly enable it. Instructions
can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+and+Aastra+Phones#DirectedCallPickup
Gareth
-Original Message-
From:
[EMAIL PROTECTED]
Hi
I am search a small information
- i use Asterisk on official IP without Nat
- My first VoIP phone are a Thomson 2030 on a NAT Network.
That's work very good.
But now, i want add a second phone, a Linksys SPA-941 on
the same network of the Thomson 2030 ...
My problems that i don't see
Hi,
I have a Polycom 501
connected to Asterisk 1.2.4 (and then connected to a VOIP provider, Unlimitel in
my case). My job requires me to attend conference calls regularly, and I
am usually there as a silent listener. Therefore, I mute my
phone.
I`ve noticed that if
I mute my phone,
On Thu, Sep 07, 2006 at 01:27:36PM +0200, Arik Raffael Funke wrote:
Hi,
has anybody had success compiling bristuff with kernel 2.6.17.11? Error
messages are below...
This is not code that is touched by the bristuff patch.
Anyway, I'd try the latest 0.3.0 bristuff patch.
Cheers,
Arik
On Thu, 2006-09-07 at 11:14 -0400, Gareth Owen wrote:
The directed call pickup functionality is turned off by default – you
have to explicitly enable it. Instructions can be found at
http://www.voip-info.org/wiki/index.php?page=Asterisk+and+Aastra
+Phones#DirectedCallPickup
I'd forgotten
Hi Guys
I too am trying to do exactly the same thing in being a provider for family
members. My Asterisk server is on a public ip, my home is behind a Watchguard
Firebox, my job is also behind a Firebox. I am using a combination of Cisco
7960, Linksys 941 and XTEN Softphone. Sometimes it works
It is in zconfig.h -- immediately before the echo cans:
/* #define CONFIG_ZAPTEL_MMX */
Just make sure it's still commented out to give my situation a try.
Moj
M.Hockings wrote:
Mojo with Horan Company, LLC wrote:
What codec are your sip phones using? We'd have a similar, though
immediate,
Noc Phibee wrote:
Hi
I am search a small information
- i use Asterisk on official IP without Nat
- My first VoIP phone are a Thomson 2030 on a NAT Network.
That's work very good.
But now, i want add a second phone, a Linksys SPA-941 on
the same network of the Thomson 2030 ...
My
Does this device allow connection to other phones besides Skype, like
Xten Xlite?
http://www.voipvoice.com/UConnect-2.html.
Compatibility with standard voip is not mentioned on their website?
___
--Bandwidth and Colocation provided by Easynews.com --
In article [EMAIL PROTECTED],
Mike [EMAIL PROTECTED] wrote:
I have a Polycom 501 connected to Asterisk 1.2.4 (and then connected to a
VOIP provider, Unlimitel in my case). My job requires me to attend
conference calls regularly, and I am usually there as a silent listener.
Therefore, I
I've just installed Asterisk using TrixBox 1.1 (previously has 1.0
installed and working).
All my sip trunks and iax trunks connect and can receive calls (there
are no phones connected to Asterisk - it's just used for incoming
automated services), but the problem is that the line is silent.
The
Hi matt,
sorry this might be a stupid question but is a bit pertinent to me,
I'd asked something similar in one of my last email regarding SMP. Do
you know if (*) is capable of making use of HT support i.e is
multi-threaded and improves performance for operations like
transcoding? Is that a
Stupid question where did you find it ?
Looked at their site downloads and under the extranet site but could
only see old versions.
Nathan.
On 07/09/2006, at 10:21 AM, Chris Dos wrote:
Well, it seems that Polycom has release new firmware 2.0.1 and
bootrom 3.2.2.
I've proceded to
Hi Dan,
Dan Austin wrote:
I ahve been using the RTP packetization patch for a while, and
its going great. I have a few questions:
That is excellent.
I always get this message:
2006-08-31 22:11:22 WARNING[1278]: frame.c:1072
ast_codec_pref_getsize: Framing not set for codec alaw, using
Polycom are analy retentive about giving out software updates.
-Original Message-
From: Nathan Alberti [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 07, 2006 10:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom new firmware
Hello,
Could anyone tell me how to install/configure H323 with Asterisk 1.2.11 .
Thanks
Wazb
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi Elpidio,I am Chandra from India. I have a doubt. I am trying to solve my problem from many days. But, I couldn't able to solve this problem. I am using Asterisk with Redhat Enterprise Linux 4.2. The problem is 5060 port is blocked. After stop my firewall (service iptables stop) also, 5060 port
Typically you have to go to a reseller who you purchased Polycom equipment from. Even then it can be tricky since they have to find away to get you the files with out upsetting Polycom.
On 9/7/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Polycom are analy retentive about giving out software
All authorized Polycom resellers will have access to this firmware and are required to provide this firmware to you. Contact the reseller you purchased the Polycom phone from. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP
Micheal,I do this with the zip version of idefisk avaliable here : http://asteriskguru.com/tools/idefisk_windows.phpI download and extract the files the run the phone and configure the settings and the speed dials, all of which is stored in the folder with the application. I then zip it up and
As far as the above is concerned I have the following:
I am using Asterisk 1.2.10, patched with this patch for 1.2.10.
I have 2 * boxes. They call each other over SIP, and I have in
sip.conf on both boxes
autoframing=yes
disallow=all
allow=g729:80
When A calls B, it sets ptime:80.
Thanks Tony. Its possible that the phone stops sending RTP stream (but it
certainly is receiving some!). How do I get Asterisk to stop caring whether
it receives RTP or not?
Yes there is a NAT between the phone the the Internet. The Asterisk server
doesn't have NAT though.
I'll try to find out
yusuf a écrit :
Hi,
you dont have to/should'nt be using different SIP ports for each
phone. Its completely not needed. Also, you dont have/need to port
forward. Just open ports 5060 and 1000-2, on the box that
asterisk is running, and on your NAT router. Dont port forward.
Then in
That
process is worse than pulling teeth!
-Original Message-From: Jessee J Holmes
[mailto:[EMAIL PROTECTED]Sent: Thursday, September 07, 2006
11:25 AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [asterisk-users] Polycom new firmware and
You've never tried to get firmware for the Cisco 7960 I take it? =) I'd rather try to write it myself then go through that again.-brandonOn 9/7/06,
Douglas Garstang [EMAIL PROTECTED] wrote:
That
process is worse than pulling teeth!
-Original Message-From: Jessee J Holmes
I think remember there is a readme on /docs that talks about
chan_h323.Check it !
Anyway you could try too at voip.info dot org.
Regards
Wasif escribió:
Hello,
Could anyone tell me how to install/configure H323 with Asterisk 1.2.11 .
Thanks
Wazb
Thanks but question!
In this folder I see:
the original Zip file i downloaded -
idefisk137.zip
addressbook.conf
idefisk.conf
hostory.txt
iaxclient.dll
Idefiskmanual.htm
idefisk.exe
Using Wordpad, I opened addressbook.conf and
idefisk.conf but saw no reference to the IP address of my
Crazy Boy wrote:
Hi Elpidio,
I am Chandra from India. I have a doubt. I am trying to solve my problem
from many days. But, I couldn't able to solve this problem. I am using
Asterisk with Redhat Enterprise Linux 4.2. The problem is 5060 port is
blocked. After stop my firewall (service
Hello Michael,
I just had both Mom and my brother up as extensions on my Asterisk pbx
using IAX2, the Cubix phone for now, but I downloaded and tried several. I
loke multiple lines, but a clean GUI is better for my family..
Oh yeah, it worked flawlessly :)
I open one port to my server
You need to MAKE a sample config by configuring your phone first, then ya
get a nice little .xml config file you can batch tweak. :) That's what I
found out.
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Thu, 7 Sep 2006,
I agree, Polycom should make this publicly available; but unfortunately, I've seen worse policies out there *cough* Cisco *cough*.The reseller shouldn't give you any hassle about it and if they do, or if you can't reach them for whatever reason (A.K.A. no email replies or phones being
I have access to the ftp server of polycom
--- Douglas Garstang [EMAIL PROTECTED] a écrit
:
That process is worse than pulling teeth!
-Original Message-
From: Jessee J Holmes [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 07, 2006 11:25 AM
To: Asterisk Users Mailing List -
But does it help ? Is it better than before ?
Do you have a good way of debugging ? (like an audio recording that i
could play ?)
Does it show something on the cli when it happens ?
Zoa
Servetas, Andrew wrote:
They recommended changing the default value of 1000 up or down
incrementally
Also keep in mind that as of right now, the latest bootrom and firmware available from Polycom (and thus your reseller) are Bootrom 3.2.2 and Firmware 2.0.1The 2.0.1 firmware is new as of a day or two and include some enhancements for buddy lists and shared presence as well as newly added secured
Tzafrir Cohen wrote:
On Thu, Sep 07, 2006 at 01:27:36PM +0200, Arik Raffael Funke wrote:
Hi,
has anybody had success compiling bristuff with kernel 2.6.17.11? Error
messages are below...
This is not code that is touched by the bristuff patch.
Anyway, I'd try the latest 0.3.0 bristuff
Great. Thanks very much
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Ellson
Sent: Thursday, September 07, 2006 2:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Softphones IAX vs. SIP, remote
I just went thru the same problem days ago and it all ended being problems with the firewall.Even if the application is listening in a given port, that doesn't mean the port is open in the firewall.try thise to see if the firewall is letting the traffic thru an specific port:iptables
2006-08-31 22:11:22 WARNING[1278]: frame.c:1072
ast_codec_pref_getsize: Framing not set for codec alaw, using
default 20
As far as the above is concerned I have the following:
I am using Asterisk 1.2.10, patched with this patch for 1.2.10.
I have 2 * boxes. They call each other over
The configuration is done in the softphone, like Nick mentions then you can tweak it with a text editor per individual.On 9/7/06, Ferguson, Michael
[EMAIL PROTECTED] wrote:
Thanks but question!
In this folder I see:
the original Zip file i downloaded -
idefisk137.zip
addressbook.conf
Not to mention the feature that the new firmware and bootrom that prevent it
from registering with the Asterisk server unless you hard code the sip settings.
Chris
Jessee J Holmes wrote:
Also keep in mind that as of right now, the latest bootrom and firmware
available from Polycom (and
Which one has video for the mac?On 9/7/06, Nick Ellson [EMAIL PROTECTED] wrote:
Hello Michael,I just had both Mom and my brother up as extensions on my Asterisk pbxusing IAX2, the Cubix phone for now, but I downloaded and tried several. Iloke multiple lines, but a clean GUI is better for my
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hey all,
I'm looking into setting up a system or two with either IMAP or ODBC
storage of Voicemail messages and wanted to hear about your experiences,
gather tips or warnings, etc, before I go diving too deep into it. Are
either of those storage
The
reseller doesn't hassle us... it just takes them several days to fulfill simple
requests.
-Original Message-From: Jessee J Holmes
[mailto:[EMAIL PROTECTED]Sent: Thursday, September 07, 2006
12:45 PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
RR wrote:
Hi matt,
sorry this might be a stupid question but is a bit pertinent to me,
I'd asked something similar in one of my last email regarding SMP. Do
you know if (*) is capable of making use of HT support i.e is
multi-threaded and
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Arun Kumar wrote:
hi
my asterisk -r shows me Most of the times Outgoing Spool Failed. Can some
one tell me why is it happening and how to solve this issue. Is it a
problem
?
You'd need to provide more information.
Does it work when you call
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ronald Wiplinger wrote:
My last update was a while back and as I remember svn trunk did not
compile and I was advised to use branches 1.2 till further notice.
Have I missed the further notice and can we use now svn trunk or is the
advice still
HUSHshout I think it was called...
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Thu, 7 Sep 2006, Blake Krone wrote:
Which one has video for the mac?
On 9/7/06, Nick Ellson [EMAIL PROTECTED] wrote:
Hello Michael,
I just
Does anyone know off hand which IAX softphone has IM
capabilities like XTEN?
Thanks
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Blake
KroneSent: Thursday, September 07, 2006 3:34 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
I do not seem to be able to get this right... after much reading and trying...
any suggestions would be much appreciated.
I have 2 ports on a TDM400 working...
now I want to bring my T100 with PRI online in the same machine...
using Asterisk 1.2.10
ztcfg is complaining see below... and I cannot
The Intel IPP based open source release of G.729 and G.723.1 have now
been updated to compile with the following versions of Asterisk:
- Asterisk 1.2.11
- Asterisk trunk - tested with SVN r 42264
The code is at the usual location:
http://www.readytechnology.co.uk/open/ipp-codecs/
If you
Hi Guys,I try to use Eyebean "Speex" Codec to Asterisk and transcoded to G729 outbound to Cisco. I receive very clear on Eyebeam, but transmit crappy to Cisco. Any clue?I get the following notices below:Sep 7 13:48:55 WARNING[5297]: codec_speex.c:278 speextolin_framein: Out of buffer space Sep 7
What do yo mean by fails?
If you don't if one party doesn't have the preferred CODEC Asterisk will
fall back to the next preferred CODEC and so on until a match is found.
Can't help you on the licensing thing though. I guess no one wants to
touch it since Digium's stance seems to be that you
I looked through the forums but could not find exactly what I needed. I need help setting up call forwarding in sip.conf, where the call forwards to PSTN number without a sip phone but just the channels in sip.conf without any hardware or softphone. Any help will be greatly appreciated.
What tools are you using for this?
I'm sure you are aware of SIPp but wondered if you had anything else?
Mark
On Thu, 2006-09-07 at 21:41 +0200, Matt Riddell (IT) wrote:
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RR wrote:
Hi matt,
sorry this might be a stupid question but is a
[EMAIL PROTECTED] wrote:
Can't help you on the licensing thing though. I guess no one wants to
touch it since Digium's stance seems to be that you should have a
license for each seat rather than a pool.
That's not enough.
You need one license per call, with no upper limit on the number of
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