And my dreamthat one day Polycom phones will support Australian Daylight
savings...
But it's only a dream.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick Baranowski
Sent: Friday, 29 April 2005 3:03 PM
To: 'Asterisk Users Mailing List -
On 04/15/05 16:22 chawki hammoud said the following:
--- Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
communications. ulaw is about 80kbps, and gsm about
28-30kbps.
I monitored the download and upload data rate during
my call using mandrake linux and it gave me 9.3 kb/s
using ulaw and 3.1 kb/s for
Is there a limit to how many voicemail boxes you can copy a voicemail
to? I have a group that has about 40 members and it only copies to
voicemail to 20 of them.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
wau thank you it works!! but,
first it says that e loop is detected,
and secondary what must I do to hand over the new workingchannel to my x-lite to use it???
DENGENSDana Olson [EMAIL PROTECTED] wrote:
On 4/27/05, Guy Boehm <[EMAIL PROTECTED]>wrote: Hello, I want to call a peer over the
wau thank you it works!! but,
first it says that e loop is detected,
and secondary what must I do to hand over the new workingchannel to my x-lite to use it???
DENGENSRichard Lyman [EMAIL PROTECTED] wrote:
Guy Boehm wrote: fputs($socket, "Channel: 6159bfb47b9\r\n\r\n");Response:
We recently had our PRI installed, we currently have 100 toll-free's
pointing to it.
I have almost everything working great but..
I have setup the first few numbers we want to use coming in from the PRI and
they work great, but..
What I want to do is setup an extension with pattern matching to
Hi !
I wanted to use Barge IN with queues. ACtually what I want to do is a SIP
user comes in a queue and then goes to a SIP agent. I want any application
that allows me to listen to the conversation between them. I can be a
supervisor extension or anything. I have used Flash
In article [EMAIL PROTECTED],
Josiah Bryan [EMAIL PROTECTED] wrote:
On Thursday 28 April 2005 11:07 am, barney wrote:
Hi there,
I`m trying to add some prefix before my local extensions, when my calls are
routed to ZAP trunk.
(i.e.: my local extension is , and i would like to send
I'm testing this strange behavior using livevoip, teliax, and
voicepulse connect. I'm calling our office phone which picks up after
two rings and plays a greeting. With livevoip and teliax I hear 3-4
rings and when the line answers I find myself a few seconds into the
initial greeting. With
Daniel Salama wrote:
This is great information. I have the following questions based on a
hypothetical scenario and some assumptions:
Based on the price of these configurations, I wouldn't even mind putting
two servers each with 2 T1s just so that I could get all calls recorded
and distribute
In article [EMAIL PROTECTED],
barney [EMAIL PROTECTED] wrote:
I`m trying to add some prefix before my local extensions, when my calls are
routed to ZAP trunk.
(i.e.: my local extension is , and i would like to send request to my
telco provider
with source phone number 55)
Hello,
Wrote some Python scripts last night to scratch an itch I was having
with Asterisk.
http://www.beerandspeech.org/cgi-bin/blosxom.cgi/tech/linux/050429a.html
http://www.beerandspeech.org/images/050429/asterisk-config.py.txt
http://www.beerandspeech.org/images/050429/asterisk-passwd.py.txt
exten = ,1,Dial(Zap/g1/5${EXTEN}/);
Thank you Josiah, but If i do that, asterisk only add prefix to extension
which i`m dialing. But that is not my goal. I need to add prefix before my
local extension:
IP_PHONE --- ASTERISK PSTN --- TDM_PHONE
ext.
Paul Hales wrote:
And my dreamthat one day Polycom phones will support Australian Daylight savings...
But it's only a dream.
Unless I am missing something, you don't need to dream about it - set it
in ipmid.cfg.
Look at the Sip Admim PDF for an explanation of:
I wrote:
I haven't tried this, but the first thing I would try is this (replace
with the extension pattern you are using):
exten = ,1,SetCIDNum(${PREFIX}${CALLERIDNUM})
exten = ,2,Dial(.)
where PREFIX is a global variable containing the prefix you want to prepend.
of
Hi friends !
Cvan anybody help me to configure asterisk with ser so that I can share the
load of the asterisk with ser server. I have tried it but my asterisk is not
showing registrations of the useragent, as given in the asterisk
wiki/asterisk+at+large. I don`t know what is the problem, but
I have the current version og mpg running. But i am geeting the same problem
even with the ringing tone. It seems to disappear sometimes
make[1]: Entering directory `/usr/src/mpg123-0.59r'
make[2]: Entering directory `/usr/src/mpg123-0.59r'
make[2]: `mpg123' is up to date.
Thanks Tony, that is exactly what i was looking for :)
-b
- Original Message -
From: Tony Mountifield [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 29, 2005 9:41 AM
Subject: [Asterisk-Users] Re: Prefix to CALLING Number ?
In article [EMAIL PROTECTED],
barney
Hi friends !
Cvan anybody help me to configure asterisk with ser so that I can share the
load of the asterisk with ser server. I have tried it but my asterisk is not
showing registrations of the useragent, as given in the asterisk
wiki/asterisk+at+large. I don`t know what is the problem, but
I did the modification that Rich explains in his email on March 23rd below.
I believe it works for me, because before this mod I was getting Ouch, part
reset... errors at least once a week, rendering * unsuitable for production
systems. After this mod, the system is running flawlessly for
Good day all
This is a error that keeps on popping up in my /var/log/messages when I
get incoming or outgoing calls on my bri card connected to 4 telco isdn
units?It is a junghanns 4 port card with the latest version of the
drivers and latest asterisk
Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for
Did you put your card in TE mode ?
To it seems you have configured your card to act like a NT but if you
are connected to bri telco lines, it should be in TE mode
check in your zaptel.conf : bri te signalling
regards
David
-Message d'origine-
De : Altus Snyman [mailto:[EMAIL
if I do a zttool it shows TE mode
On Fri, 2005-04-29 at 12:14, David Masure wrote:
Did you put your card in TE mode ?
To it seems you have configured your card to act like a NT but if you
are connected to bri telco lines, it should be in TE mode
check in your zaptel.conf : bri te
The problem may then originate from the NT of your telco
-Message d'origine-
De : Altus Snyman [mailto:[EMAIL PROTECTED]
Envoy : vendredi 29 avril 2005 12:21
: David Masure
Cc : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users] bri error
if I
and I have
signalling = bri_cpe_ptmp
On Fri, 2005-04-29 at 12:14, David Masure wrote:
Did you put your card in TE mode ?
To it seems you have configured your card to act like a NT but if you
are connected to bri telco lines, it should be in TE mode
check in your zaptel.conf : bri te
Version 0.110 - 29. April 2005.
Completely rebuild with .NET Version 2.0 BETA 2.
IPS now has MDI (Multiple Documents Interface) meaning that you can have
many program open at the same time, they will all update in real-time.
Hotel/Call Shop Billing module.
Status of Agents can be monitored on
Hi
I've successfully installed Asterisk-1.0.7,
I've successfully installed Openh323 gatekeeper but not registered to Asterisk
and so
I've install PWlib v1.5.2 and Openh323 v1.12.2 libraries
Now i try to install asterisk-oh323-0.5.10 :
-edit Makefile inside the asterisk-oh323-0.5.10 directory and
Hi Pros,
I`m new to Asterisk Getting following errors on my * :
-- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to
Just for future reference, I found the answer - I enabled Symmetric RTP: on
the Advanced SIP page.
Chris Mason
www.anguillaguide.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Mason (Lists)
Sent: Thursday, April 28, 2005 6:22 PM
To:
Hi,
Yes, I understand that 'Ouch.. etc' comes
frommpg123. So I'm not loading musiconhold to avoid this problem (It
happened when exiting asterisk, nothing to do with the core
dumped).
And now, IAX is still crashing and turned
asterisk down with Segmentation fault everytime I make an IAX
Hi friends !
Cvan anybody help me to configure asterisk with ser so that I can share the
load of the asterisk with ser server. I have tried it but my asterisk is not
showing registrations of the useragent, as given in the asterisk
wiki/asterisk+at+large. I don`t know what is the problem, but
I'm using dtmfmode=inband with Sipura=3000 when I dial an internal
extension most of the time the first digit is missing and I get an
invalid extension message.
Could it be dtmf problem or SIP?
On the spa3k, I've use dtmf tx method = auto. In sip.conf, no dtmf
entries at all (uses
If you go to the fcc.gov website and search for CALEA there is around 7
documents that come up for April 27 2005. I believe I remember reading it
one of those documents.
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Hi friends !
Cvan anybody help me to configure asterisk with ser so that I can share the
load of the asterisk with ser server. I have tried it but my asterisk is not
showing registrations of the useragent, as given in the asterisk
wiki/asterisk+at+large. I don`t know what is the problem, but
Hi,
I see others have had this problem. Is there a solution ?
I have a BRI, using zaphfc. If I enable debugging so:
bri debug span 1
and then make an incoming call I can see that the DNID info is
definitely provided by the PSTN - Here's proof:
Called Number (len= 7) [ Ext: 1 TON:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] wrote:
Hi
I've successfully installed Asterisk-1.0.7,
I've successfully installed Openh323 gatekeeper but not registered to Asterisk
and so
I've install PWlib v1.5.2 and Openh323 v1.12.2 libraries
Now i try to install asterisk-oh323-0.5.10 :
Guys
I have a problem getting a TDM400P card to go.
It has 4 FXS ports (green modules) and I get this error:
[EMAIL PROTECTED] root]# ztcfg -v
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default)
Hi there
I don't know if this utility is available anywhere at the moment but I
thought id ask you guys if you know of one
What I would like is a way of adding a field to my cdr records (either the
Master.csv or a destination mysql table) for cost ! based on some sort of
config file (or table)
Hi all.
I am starting to develop a voip solution with asterisk and sip and 100
telephones with SIP .
I need the following features.
calls group
blind transfer
attended transfer ( supervised )
pickup groups
voicemail
record call
the attended transfer feature is very important, cause is an
The homepage http://sipsak.org contains some examples. If you need help with
special cases drop me a line.
Regards
Nils Ohlmeier
On Friday 29 April 2005 02:54, Anton Krall wrote:
Can you send some command line examples on how to use it?
Thx!
|-Original Message-
|From: [EMAIL
I'm testing this strange behavior using livevoip, teliax, and
voicepulse connect. I'm calling our office phone which picks up after
two rings and plays a greeting. With livevoip and teliax I hear 3-4
rings and when the line answers I find myself a few seconds into the
initial greeting.
For the record archives, there are apparently about eight different
revisions of the TDM card (observed via pci id revisions in driver code),
and this mod only impacts one of apparently multiple problems. Since
digium is very tight lipped about problems, there is a high probability
that other
Hi !
I am using queues with MOnitor Application but the thing is that Iwant to
save the files starting with the Answering agent name. I have tried a lot
of things but nothing seems to work. If i put Monitor application on top
of dialing the agent then as soon as agent picks up the recording
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
Does someone knows if the next release of Asterisk (1.0.8?) will have Realtime
support and when we will have the next Asterisk release
with Realtime features?
Thanks in advance.
- --
Rodrigo P.
Anton Krall wrote:
Guys
I have a problem getting a TDM400P card to go.
It has 4 FXS ports (green modules) and I get this error:
[EMAIL PROTECTED] root]# ztcfg -v
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart
Why would you use gateways and PRI's when several of the major carriers
(ATT, Global Crossing, etc.) also have products that can interface
directly with SIP for the same per minute cost?
We have a multisite Asterisk call center application and are routing all
calls over private VPN to one central
I have a problem getting a TDM400P card to go.
It has 4 FXS ports (green modules) and I get this error:
[EMAIL PROTECTED] root]# ztcfg -v
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart
I am having an issue with the asterisk system not responding to dialed
numbers during an active
call. I'm not even sure where to look, zapata.conf? sip.conf? or the phone
config? and worse I
don't even know what Keywords to search for.
Tim
___
On 4/29/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi !
I am using queues with MOnitor Application but the thing is that Iwant to
save the files starting with the Answering agent name. I have tried a lot
of things but nothing seems to work. If i put Monitor application on top
of dialing
Hi @all,
I have following hardware setup:
standalone analog fax-machine - DeTeWe TA33clip a/b adapter -
OctoBRI S0 NT-Mode - Digium TE410P - german PSTN
Software: Debian unstable binary packages
ii asterisk 1.0.7.dfsg.1-2 open source
Private Branch Exchange
Guys
I have a problem getting a TDM400P card to go.
It has 4 FXS ports (green modules) and I get this error:
[EMAIL PROTECTED] root]# ztcfg -v
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02:
FXS Kewlstart
The disk array would be the only expensive add on, more than a normal
asterisk system. It all depends on how important voicemail is in your
application, although there are cheaper alternatives (NFS for example,
but then your NFS server becomes a single point of failure, depending on
the disk
Zttool shows nothing inside thebox.
I tried removing the x100 cards, moving the tdm card around, disabled all
usb and unnecessary stuff still, kudzu when booting up shows the card and
the card shows up on /etc/sysconfig/hwconf but I wonder why it shows 2 of
these and I only have 1 tdm400p card
This is a DTMF issue,
You must adjust this on the especific channel conf file.
For example, ia sip phone cannot dial any number during an active call, you must see sip.conf and the config in your hardphone or softphone.
Ismael.
Tim Touhsaent [EMAIL PROTECTED]
Enviado por: [EMAIL
Is this a SIP phone?
I had to upgrade the firmware on my SIP phones to alleviate this. It seems that
the phone would actually disable it's own keypad after dialling.
Ian
[EMAIL PROTECTED] 29/04/2005 09:16
I am having an issue with the asterisk system not responding to dialed
numbers during
Zttool shows nothing inside thebox.
I tried removing the x100 cards, moving the tdm card around,
disabled all usb and unnecessary stuff still, kudzu when
booting up shows the card and the card shows up on
/etc/sysconfig/hwconf but I wonder why it shows 2 of these
and I only have 1 tdm400p
Anton Krall wrote:
Zttool shows nothing inside thebox.
I have had similar problems with a TDM400 and CERTAIN Motherboards which are
PCI 2.2 but the TDM400 is not seen, in my case, AT ALL
The one I have reports it as an E/F but the silk-screen clearly says H, Digium contends
there is no problem
Has anyone had any experience (good or bad) running Asterisk under
VMWare ESX server on a blade chassis? This application will (fairly
obviously) not include Zap channelsactually, it will be SIP-only.
Please feel free to contact me off-list and I'll summarize for the list
later.
Daryl G.
If price would truly not an option just get one of the Signate Telephony
5000 servers(http://www.signate.com/pbx.php) They are about $18,000 and
allow you to have upto 5000 SIP streams go through it. You could have that
be your gateway and do the SIP-IAX through that machine and scale upto 100
T1s
Rich Adamson wrote:
For the record archives, there are apparently about eight different revisions of the TDM card (observed via pci id revisions in driver code),and this mod only impacts one of apparently multiple problems. Since digium is very tight lipped about problems, there is a high
Kerry,
Thanks for the reply but we are looking for someone in the Chicagoland area.
Regards,
Jon Dahl
SKTY Trading, LLC.
From: Kerry Garrison [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: 'Asterisk Users Mailing List -
Yes, I have a snom 190. I'm gonna check out the dtmf signalling now. thank
you for the quick responces.
Tim
- Original Message -
From: Ian Pattison [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 29, 2005 9:44 AM
Subject: Re: [Asterisk-Users] need help
Is this
I would like to record two months of calls.
The call center does not have a huge volume, probably like 60 calls a day and
average about 15 min a call. I am using a quad port e1 card from
digium. i would like to record the calls on a seperate server than the one
running asterisk to avoid any
Dear All,
I'm using CVS-HEAD 06/04/05 with Realtime, and at present, its working fine
generally. However, I'm facing a problem that I find it strange and would
like to seek your kind advise.
I'm using Firefly 1.9.8 build 3945 and I realise that when I set qualify to
yes, then then Asterisk will
On Fri, Apr 29, 2005 at 12:23:48AM -0500, Brian Capouch wrote:
Drat. Perl screams bloody murder if you try to just set its SUID bit,
which of course is dangerous as hell.
The perl-suid is *not* simply a version of perl with the suid bit set
but rather a helper binary which allows perl to
Zttool shows nothing inside thebox.
I tried removing the x100 cards, moving the tdm card around, disabled all
usb and unnecessary stuff still, kudzu when booting up shows the card and
the card shows up on /etc/sysconfig/hwconf but I wonder why it shows 2 of
these and I only have 1 tdm400p
Please can anyone help me with my quadbri card
I am desparate L
I compiled the bristuff drivers and then I do
--
Modprobe zaptel
Insmod qozap.ko
Ztcfg
The it complains it cant find
You need something like this ??
exten = _0.,1,SetVar(CALLFILENAME=${CALLERIDNUM}-${EXTEN}-${TIMESTAMP})
exten = _0.,2,Monitor(wav,${CALLFILENAME},m)
exten = _0.,3,Dial,SIP/[EMAIL PROTECTED]
and mount another server with NFS or SAMBA on /var/spool/asterisk/monitor
That would be the job.
Sjaak
I
60 calls a day is nothing. I'm sure your Asterisk box can handle it
with the standard Monitor command.
I've recorded many calls, 8+ hours straight and I'm on a crap old
Pentium 3 633MHz system.
What exactly do you fear will happen if you record on the Asterisk box?
--
Dana
On 4/29/05, Steve
http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-04-187A1.pdf
http://www.wi-fiplanet.com/voip/article.php/3390671
http://www.cybertelecom.org/voip/Fcc.htm
(scroll down)
and of course:
FCC To Require 911 for VoIP
http://www.newsfactor.com/story.xhtml?story_id=33733
- Original Message
smells like udev. Checkout README.udev in the zaptel directory.
On 4/29/05, Sander [EMAIL PROTECTED] wrote:
Please can anyone help me with my quadbri card
I am desparate L
I compiled the bristuff drivers and then I do
--
I was wondering if anyone is working on graceful failure for chan_zap?
Let me explain the situation. We are using a T100P and TDM400P (4 FXS
for fax). There was a major power outage and asterisk went down after
the UPS (not a graceful shutdown -- my fault, no apcupsd running).
As soon as
I think you will find AMP is about to implement a multi tenant solution.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Niemantsverdriet
Sent: Friday, April 29, 2005 1:15 AM
To: Asterisk Users Mailing List -
Hi all,
Does anyone know of a way to setup global var using the manager interface.
Basically I want to be able to have multiple manager clients login,
however in a sort of master slave scenario. So the first client that
logs in, sets a global variable which tells subsequent clients at
least
Hi,
Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs ,
one with 15 channels and the other with 15 channels;
Is there a sort of E1 multiplexer devise that allows me to plug in one hand the
E1 port of the Digium card and on the other hand the two PABXs? In this same
No udev installed on my system :( so that does not help me
Thanks anyway
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Michael Bielicki
Verzonden: vrijdag 29 april 2005 17:18
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re:
It is a critical system and located overseas with no technical people
onsite. Logic dictates that changes be made with a light footprint.
- Original Message -
From: Dana Olson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Thank you, For the responces i had dtmfmode=inband
when rcf2833 was the proper setting. I feel retarded that i missed that, but it
happens. thanks again
Tim Touhsaent
- Original Message -
From:
[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial
Hi all
I have installed version 0.9 against a supplier SIP and have put a AVM B1
like backup.
The reception of calls works perfectly, but with himself not to make calls
to ISDN card.
It is possible that this configuration works whith [EMAIL PROTECTED] ? or I am
mistaken.
AVM card is
On April 29, 2005 11:22 am, Jeb Campbell wrote:
As soon as power came back, the server started. However when it loaded
wcfxs, port 3 on the card failed the tests (I assume from the module not
being unloaded before power off). Because this one port failed the
test, chan_zap failed to load and
Rodrigo P. Telles wrote:
Does someone knows if the next release of Asterisk (1.0.8?) will have
Realtime support and when we will have the next Asterisk release
with Realtime features?
Where is your failure? I don't see anything. The next stable release of
asterisk will be 1.2 and it will have
I am running on usermodelinux
Itamar Reis Peixoto
+55 (34) 3238 3845
e-mail : [EMAIL PROTECTED]
http://vps.ispbrasil.com.br --- servidores linux
Has anyone had any experience (good or bad) running Asterisk under
VMWare ESX server on a blade chassis? This application will (fairly
obviously) not
yes, some multiplexer allows that, but they're quite expensive
compared to another E1 card for asterisk.
I think you'll need at least 1k $$$ for a such splitter.
Matteo.
Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto:
Hi,
Assume I have one E1 digium card to which I
Hi,
when I dial my voicemenu the menu voice is always cutted so that i only
hear 'word from password.
What do i have to configure so that i hear the full text from the beginning?
thanks, Kib
___
Asterisk-Users mailing list
On Friday 29 April 2005 12:12 pm, Kib Eki wrote:
Hi,
when I dial my voicemenu the menu voice is always cutted so that i only
hear 'word from password.
What do i have to configure so that i hear the full text from the
beginning?
thanks, Kib
You might try inserting a Wait in your menu
Wouldn't introducing Samba into the mix be even worse?
I would think it would add more processing power and network use to be
constantly writing over the network as opposed to recording on the
same box.
If it's such a critical system, it should have the power to do that,
but that's not the
Sure.
I setup a small lab on a machine with 4 T1s and 36 agents logged in.
The system was configured to Monitor all outbound calls as well as
monitor all calls distributed by Queue app (monitor-format setting in
queues.conf).
When recording to local disk, everything was working fine. Agents
Andrew Kohlsmith wrote:
It has nothing to do with not being unloaded; I've seen the wctdm driver fail
to detect modules correctly. Run it again and it works just fine. Some kind
of minor tweak is in order, I believe.
As an interim solution, your asterisk starup script should try to unload any
Well, I don't think I'm ready to spend that much money :)
I understand your point regarding that load depends on usage.
SIP_Agents are simply agents answering calls. Average call length would
be about 8 minutes. During some of these calls (maybe 25%), agents will
conference the call (PSTN
Greetings,
I have two machines. One is a P3 Dell Dimension 4100, the other is a
PowerEdge SC420. Both are running Asterisk 1.0.7, the PowerEdge has a
TE405P card in it, the Dimension has a Digium X100P present (although
not modprobed). Each machine has mpg123 0.59r loaded, and is using
the
Hi,
I am doing some testing with asterisk using Cisco IP Phones 7960's and
EyeBeam. I have canreinvite=yes on all my devices but the media still
goes through the asterisk box. Does it mean that Cisco and Xten do not
support re-invites? If yes can you recommend SIP phones or adapters
that support
On April 29, 2005 12:38 pm, Jeb Campbell wrote:
While I like the idea (and will look into it -- might need a wait, etc),
as I said in original post, unloading and reloading did not fix the
problem. It took a clean shutdown (unload and restart) to fix the problem.
Hmm; that is odd...
So
Upgraded one of my asterisk servers to the latest cvs head version last
nigh now I get one way audio on IAX2 channels when calling other
asterisk servers. Anyone seeing the some problems?
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Ok, this is probably stupid question of the week. I have
exten = 888,1,whatever
exten = 888,n,UserEvent(Event|Data)
exten = 888,n,Hangup
If I asterisk -r, when I dial the 888, I see Userevent appearing in the
console.
However, if I telnet to the * manager using a name and password that has
the
There isn't a specific command in the manager API itself to do it.
However there is a CLI command and you can use the manager command
action to get the information. Below is an example, you will need to
parse the response part to see who is connected.
Action: Command
Command: show manager
Daniel,
Thanks alot for this post. You were right on time with valuable
information.
Thanks again,
Steve
- Original Message -
From: Daniel Salama [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 29, 2005
Do you get 2-way audio that sometimes drops off to 1-way audio then picks
back up as 2-way? (Thats what I see)
Not sure if my problem is a lost packet issue as I am sending IAX off net.
Duane Cox
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
David Josephson,
Not off-base, but you haven't made it all the way home yet. This is
another layer of the puzzle, and again we are not talking about apples
and apples here. Circuit switched means that there is a (real or
virtual) circuit that takes data on an input port and delivers it to
an
Guy Boehm wrote:
wau thank you it works!! but,
first it says that e loop is detected,
and secondary what must I do to hand over the new working channel to
my x-lite to use it???
DENGENS
Richard Lyman [EMAIL PROTECTED] wrote:
Guy Boehm wrote:
fputs($socket, Channel:
I'm using sip-tester you should try it
gnuws:~# apt-cache search sip-tester
sip-tester - a performance testing tool for the SIP protocol
gnuws:~#
On Fri, 2005-04-29 at 14:14 +0200, Nils Ohlmeier wrote:
The homepage http://sipsak.org contains some examples. If you need help with
special cases
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