Hi Ken,
Can't seem to find it anywhere, and my cisco login works, but says
there's no longer any downloads available for the ATA186.. anyone know
where I could find the MGCP version of the firmware via download?
Log in. From the main page, click the dropdown list for
Downloads and select
Jean-Michel Hiver wrote:
Hi List,
I have seen this:
http://www.convergence.com.pk/iax2/trunked.html
According to this table, using trunking, you can have 16 channels with
171.7 kbps bandwith using g.729 + IAX2 trunking? Sounds too good to be
true...
Any comments on this?
If I'm reading the
Hi
I am currently running [EMAIL PROTECTED] version 0.9 and have a few questions,
which i hope someone on this list might be able to answer.
1) I am trying to setup incomming fax support, but however i never manage
to receive the faxes, getting a signal 15. As per handbook, there isn't
too much
On Mon, 25 Apr 2005, Andrew Kohlsmith wrote:
It has absolutely nothing to do with what economically suits them best --
it
has everything to do with the fact that when you buy a clone X100P you DO NOT
KNOW what you're getting. The chipset may be the same but as you can clearly
see from
Joseph wrote:
Can anybody explain me why IAX is called proprietary protocol?
In some places IAX is refereed as open protocol.
How can proprietary protocol be open protocol?
Since the source code is available to anyone and GPL'ed it is an open
protocol.
However it's not a standard and there is
Hi Chris,
I've had A LOT of experience with the cheap X100Ps in the last few weeks. I
myself bought two of them off ebay. $6.95 special! I also had problems with
them and within the past 12 hours have replaced them with a TDM22B that so
far(1 phone call) has worked great. I would suggest turning
Does anyone know of a Linux SoftPhone that will play nicely with ESD?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hi Peter!
FYI: Yesterday i put Asterisk between a Hicom 350E and a Telekom Austria
(TA) PRI. Both use TON=unknown for called number, but Hicom always uses
TON=international for calling number whereas TA uses a dynamic TON for
calling number. Thus, for incoming calls (PSTN-PBX) the presented
As mentioned yesterday, i made an attempt to write documentation to get
NAT + SIP to work on http://www.asteriskguru.com/natut.php
If you send me the info for those phones, firewalls i will include them.
(I was planning on adding some Linux/BSD firewall rules but i dont have
a pix,).
/Z
Irakli
Hello Tomek,
I also got a diva pci 2.02 card, but although the kernel sees the
incoming calls, asterisk refuses to answer. Did you have this issue at
all?
The kernel seems to be denying the call...
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I would also be interested in a multi-port ata that supports iax.
The only single port ata I know of (besides the iaxy) that supports
iax is the PA168 from china.
cheers
Clive
On 27 Apr 2005 at 11:15, Rod Bacon wrote:
What sort of price are they asking for a 4-port gateway?
- Original
I've been there.. the page comes up with There are currently no files
for this type.
:(
k
Stewart Nelson wrote:
Hi Ken,
Can't seem to find it anywhere, and my cisco login works, but says
there's no longer any downloads available for the ATA186.. anyone know
where I could find the MGCP
On Wed, 27 Apr 2005, Dan Morin wrote:
To expand upon my original question, does anyone know of any devices
that would make connectivity between the Panasonic system and Asterisk
possible? What are opinions of using FXS ports in Asterisk going into
to CO ports on the PBX? Or if I'm putting
So now that they are done how about you post the files for us? Share the
wealth.
Mark
Will be happy to do so once macro refined a little, but it is rather long
(about 600 lines) and I thought long posts were bad manners.
Otherwise this will be odne by the end of the weekend/
Paul
Gregory Wiktor - ADCom Corp. wrote:
Hello Tomek,
I also got a diva pci 2.02 card, but although the kernel sees the
incoming calls, asterisk refuses to answer. Did you have this issue at
all?
The kernel seems to be denying the call...
if you see the calling in the systlog, that's 98% of success :)
Jason Leach wrote:
About every 24-48h the Zaptel FXO port crashes. If I pick up my phone
and try to make a call on the FXS port I get a hissing and squealing
sound. Seems to be right where Asterisk makes the bridge. Also
Asterisk does not answer an inbound call on the FXO port; does not
even
It seems that there areoccasional problems
with files generated by soxmix utility.
The Asterisk console would show the following
message:
soxmix: Overriding output size to bytes for
compressed data
soxmix: help! internal inconsistency - data_written
12156 gsmbytecount 12155.
When trying
Hello Tomek,
Previously I did get asterisk to see the call, but not currently.
This is in the usa, so my msn is a 7 digit number.
The kernel is saying the following:
Apr 28 03:46:17 localhost kernel: isdn_net: call from 8005966511,1,0 -
2781980
Apr 28 03:46:17 localhost kernel: isdn_net: call
Massimo wrote:
Hi,
I successfully installed zaptel,libpri,asterisk and qozap in a Suse 9.2.
I removed the old modules loaded as default by Suse.
Now I'm triying to load qozap.ko but I receive this error:
Did you do the install with bristuff-0.2.0-RC8a ?
Works nice on debian, I guess it will be the
Gregory Wiktor - ADCom Corp. wrote:
Hello Tomek,
Previously I did get asterisk to see the call, but not currently.
This is in the usa, so my msn is a 7 digit number.
The kernel is saying the following:
Apr 28 03:46:17 localhost kernel: isdn_net: call from 8005966511,1,0 -
2781980
so this is your
Hello Tomek,
When I call my second msdn, I get the following:
== Starting Modem[i4l]/ttyI1 at incoming-isdn,2781984,1 failed so
falling back to exten 's'
-- Executing Answer(Modem[i4l]/ttyI1, ) in new stack
somersvoip*CLI Apr 28 04:04:33 localhost kernel: isdn_net: call from
8005966511,1,0 -
Gregory Wiktor - ADCom Corp. wrote:
Hello Tomek,
When I call my second msdn, I get the following:
== Starting Modem[i4l]/ttyI1 at incoming-isdn,2781984,1 failed so
falling back to exten 's'
-- Executing Answer(Modem[i4l]/ttyI1, ) in new stack
somersvoip*CLI Apr 28 04:04:33 localhost kernel:
Hello all,
I just instaled an AVM C4 card on my asterisk to connect it to the PSTN and send or revibe calls using it.
I can make calls perfectly, two calls over the same port at same time.
The problem appears when a call arribes. CAPI seems to answer it and pass it to asterisk and the call
Luki wrote:
Robert,
It looks like you're dialing 447733322998, 44 for UK, then the area
code, etc. I have sipgate.de setup to dial local numbers (any German
number) as 0+AREA CODE+NUMBER. Always dial the area code, even if you
sipgate number is in the same city. For international numbers you need
In capacity planning for production Asterisk servers it is essential to
have an accurate statistical picture of the utilisation of finite
resources such as disk space, CPU utilisation, B channels on PRIs, and
G.729 codec licences.
The first two have well defined measurement tools.
The last two
Sebastian Atala wrote:
Hi,
Someone knows how can I register my Asterisk to a gatekeeper using
zone parameters?
I'm using asterisk 1.0.7 and oh323 0.6.5.
I'm trying to register to a gatekeeper in another network and I can't reach
this with a broadcast.
Zone is the name who Cisco call the GK
I have bought some Wildcard X101P and Generic Clones for my Asterisk
PBX. Now I can place and get calls through the lines/channels.
Everything is okay but the problem is when I call outside through our
PSTN line, after few minutes the connection breaks down. The same
thing happens in case of
hello,i'm a naw asterisk user i've configured my 2 sip phones and they can place calls .i,ve also fxo card and i've configured channel ;now it's possible to recieve analog calls with my sip phone but i want to make call with my sip phone to analog it's possible?
when i dial a number my sip phone
hello,i'm a naw asterisk user i've configured my 2 sip phones and they can place calls .i,ve also fxo card and i've configured channel ;now it's possible to recieve analog calls with my sip phone but i want to make call with my sip phone to analog it's possible?
when i dial a number my sip phone
hello
i successfully installed asterisk on fedora core 3 and all what's in the
check list plus the ACTOS gui and asterisk manager but i used actos to
configure my cisco ip phones and dial/receive calls through sip.
my problem is i need to configure H323/Fax in asterisk to catch H323/Fax
from
Hi Paul,
I had the same situation - I had a 7940 with only the callmanager
firmware but would have much rathered SIP. You need to have a support
contrace with Cisco to be able to download the firmware from their website.
Thankfully the support contract only costs about $9 for the year - I was
I have a couple of Cisco 7910's and I'd like to get them working with
asterisk. I have two X101P wild cards installed and they are
functioning well (Other two cards showing Red Alarm after few minutes
conversation).
I have configured foure cisco 7960 with SIP and they are working fine
with *.I
i have the cisco 7940 7960 phones and both i can use sip with asterisk?
how can i help?
On Thu, 2005-04-28 at 10:48 +0100, Derek Conniffe wrote:
Hi Paul,
I had the same situation - I had a 7940 with only the callmanager
firmware but would have much rathered SIP. You need to have a support
--- Matt Riddell [EMAIL PROTECTED] wrote:
In order to try and confirm this, see if small
packets get loss.
There was a packet loss in all the codecs i used, but
it's hard to tell whether the packet percentage loss
is gretaer or less at different codecs and how that
help in solving the
hello Asterisk-Users
I created just for fun a rss feed for Asterisk-Users and Asterisk-Biz list
First for myself but if it is usefull for you can use it if you like.
But some questions
Is it allowed ?
If you need add-on's please let me know.
When many people will use it I need to generate a
On April 27, 2005 01:25 am, Jason Leach wrote:
hi,
I have one of the latest versions of Asterisk CVS' (1.0.7-x) and the
accompanying Zaptel drivers. The zaptel drivers are for my TDM400P w/
1FXS and 1FSO. It all runs on CentOS 4.0 and a Dell Precision 410 w/
Dual PIII 700Mhz CPUs.
About
Ok... so I can safely use a provider who uses G729 or G723 to provide
me with VoIP termination without either A) having a connectivity
issue, or B) having to pay trans-coding license costs?
If so... what do I need to do to get asterisk to use G723? Just set it up?
Am I going to have an issue if
For instance.. when I try to use G723.1 on my phone (and just call in
from my PRI line) I get:
Unable to find a path from g723 to ulaw.
Unable to find a path from ulaw to g723.
No path to translate from Zap/1-1(68) to Sip/201-80c7(1).
Same things happens if I call in on my current provider's
Hi,
After a kernel downgrade to 2.6.10 instead
of 2.6.11.7 its working correctly.
Sincerely,
Cyrille
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Cyrille Demaret
Envoy: jeudi 28 avril 2005 0:00
: 'Asterisk Users Mailing List - Non-Commercial
Here's the output of 'sip show users':
*CLI sip show users
UsernameSecret Accountcode Def.Context ACLNAT
502 1234 internalNo RFC35
501 1234 internalNo RFC35
If you need more info, just let me
Jean-Francois Theroux a écrit :
Here's the output of 'sip show users':
*CLI sip show users
UsernameSecret Accountcode Def.Context ACLNAT
502 1234 internalNo RFC35
501 1234 internalNo RFC35
If
Has anyone ever been able to fix this NMI power issue that the Dell's
have with the TDM cards? Basically locks the machine up when trying to
bring up the module.
Matt
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Jean-Francois Theroux wrote:
Here's the output of 'sip show users':
*CLI sip show users
UsernameSecret Accountcode Def.Context ACLNAT
502 1234 internalNo RFC35
501 1234 internalNo RFC35
sip show
On Thu, 2005-04-28 at 10:33 +0400, Jean-Michel Hiver wrote:
Joseph wrote:
Can anybody explain me why IAX is called proprietary protocol?
In some places IAX is refereed as open protocol.
How can proprietary protocol be open protocol?
Since the source code is available to anyone and
Ok. That's different. Here's the output of 'sip show peers':
*CLI sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
502 172.16.1.202 255.255.255.255 5060 Unmonitored
501 172.16.1.201 255.255.255.255 5060 Unmonitored
Eric
I wanted to know if there is some way i can restrict the number of
agents logged into one SIP extension. I usually find 2 or 3 agents
logged on to a single extension.
Can someone help me in this regard
___
Asterisk-Users mailing list
Hi all,
Can anybody recommend a good 1-800 number provider?
Thanks
Sean
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
pin reversal should not be an issue as the silicon labs chips on the
TDM card handle tip ring in either case.
I think you will find it is pin reversed.
So flip the RJ45 Over
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Stefan de Konink wrote:
On Wed, 27 Apr 2005, Joseph wrote:
How can proprietary protocol be open protocol?
If the protocol is fully documentated and this documententation is
available to anyone you can speak of a open protocol. It is not an open
'standard', because it is only supported by Digium,
Hi!
Thank you for answering my mail.
I think my mail wasn't exactly enough.
I want to use dialparties.agi in my own dialplan. I've tried it already,
but it don't work yet. I tried it the following way:
The users are dialing a number, this number is passed to a macro, which
calls dialparties.agi
I have a used Adtran 600 with 2 X Quad FXS, 2 X Dual FXS/DualX.3 and 3 X
Dual FXO/Dual X3 modules.
I have a Sangoma A101 Kit with RJ45 cable.
Installed in my working Asterisk system is a Digium TDM22B 2 x FXS,2XFXO
card.
I would like to replace the Digium card with the Adtran unit, can anyone
Hello,
we have setup a queue with a couple of agents, all of which are joining
in via CallbackLogin. 1 out of 10 calls coming into the queue will get
hung-up upon as soon as the agent picks up the phone.
We are running 1.0.6 bristuff RC7k (single HFC-card). SIP phones, ATAs
and outside mobile
Is someone running mgcp firmware with asterisk?
I need to verify the phone issues
Thanks.
Sergio
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
my problem is i need to configure H323/Fax in asterisk to catch H323/Fax
from the gateway and route it as t38/fax to another pbx server i
installed on windows.
how can i configure, route and convert the faxes?
___
Asterisk-Users mailing list
Thank you both for the insight. The original problem was that the voice
mail system returned a no mailbox found error since the query was looking
for a mailbox in the default context and I had defined them in other
contexts, in my case, from-sip and analog-phones. It seems I am
confusing
Also, shortly after cisco purchased linksys (and with a little journalistic
help), many of the past problems with their software finally 'surfaced'
and were addressed. Given cisco's long history (internally) to standards
and quality, I'd have to take some sizable bets on product improvement
as
Hi
I've a problem with the registration of the openh323gatekeeper.
First I've downloaded and installed the pwlib and openh323 libraries
successfully.
Then
I've downloaded the package openh323gk.tar.gz,executed the binary file,
but the gatekeeper is not registred on asterisk!
Then I've also
I have a TDM400 with 4 fxo ports installed in my
IPPBX box. When I call in my IPPBX through this card
and after it answers I hangup, IPPBX still keeps going
to timeout. It cannot recognize the hangup signal from
PSTN.
Anyone knows the solution.
Past problems with disconnect
Julio Arruda wrote:
But there are royalties or something like that ?
I understand that proprietary protocols CAN be published, but what make
them proprietary is the requiremenf or royalties or at least a 'ok' from
the owner ?
Obviously IAX/IAX2 does not and should not require licensing fees.
I
In your case, where you will need the license is on the box that your
phones register to. For exampe, when someone checks voicemail,
encoding takes place, therefore you need a license.
Look at it this way:
[g729 provider] -(SIP or IAX)--- [g729 asterisk server]
- no license
Paul wrote:
Do you still have that image for the 7960? I bought a 7940 on ebay and it
doesn't have the SIP firmware. I can't find it anywhere but Cisco's website
and they require that I have an account with them. Did you happen to save
that binary file?
Cisco charges for the SIP firmware. You can
Not that I know of I am a Cisco partner and the Category 1 contract
is still at least half that or less.
He was talking about the SIP-license... Not the SmartNET. If you have a
SmartNET, you CAN download the SIP load but to use it, you need the
license.
I think that's the
Hi,
What do you have h323 or oh323 ,
(open h 323)
I think you have the latest.
You must use the SPECIFIC files.
Check http://www.inaccessnetworks.com/projects/asterisk-oh323
Also PATCH the file BEFORE compilation
It should run then.
Good luck
Regards,
Shaoul Jacobson
Senior VoIP Consultant
So
[g729 provider] -(SIP or IAX)--- [g729 asterisk server]
This is how I'd be setup.. actually more like this:
[g729 provider] --(sip) [g729 asterisk
server](sip)---[g711 sip phone client].
So... if I understand this correctly.. I *would not* for *any* reason
I'm trying to configure an asterisk box with two cards.
Incoming calls are working fine with two ISDN cards, however, I am able
to make outgoing calls only through the first card.
exten = _0.,1,Dial(Modem/g1:${EXTEN:1})
exten = _9.,1,Dial(Modem/g2:${EXTEN:1})
If I try to use the second card,
Hi there,
I`m trying to addsome prefixbefore my local
extensions, when my calls are routed to ZAP trunk.
(i.e.: my local extension is , and i would like to
send request to my telco provider with source phone number
55)
Is there any way to do this ? I just know toadd prefix
On Thu, 2005-04-28 at 11:03 -0400, Matt wrote:
So
[g729 provider] -(SIP or IAX)--- [g729 asterisk server]
This is how I'd be setup.. actually more like this:
[g729 provider] --(sip) [g729 asterisk
server](sip)---[g711 sip phone client].
So... if I
I searched for the mailing list guidelines on google and couldn't find them.
I apologize in advance if this is not the appropriate list.
My company is moving their office and we have decided to use VoIP for our
phone solution. We will be using Cisco 7960 phones powered by a Cisco 3560
switch.
exten = 2201,1,agi,notify.agi
exten = 2201,2,Dial(Zap/9,20)
exten = 2201,3,Answer
exten = 2201,4,Wait(1)
exten = 2201,5,Voicemail(u${EXTEN})
exten = 2201,6,Hangup
exten = 2201,105,Voicemail(b${EXTEN})
exten = 2201,106,Hangup
I realize that I did not need to use the EXTEN variable, since
I'll gladly pay $10 a license... I'm all for supporting digium...
however, I was under the impression that there was also some huge one
time fee of like $2,000 or something. I guess I was wrong... ok now
bad..
So I purchase the license from digium... then what happens/what needs
to be done on
We do a good amount of remote work if that isn't a problem for you. We can
reconfigure the entire system and have it ready to drop into place. If the
job is big enough it might warrant a visit during installation but that isnt
always the case.
Kerry Garrison
Tech Data Pros
Hello,
I can't use IAX with my last
CVS-NHEAD-04/28/05-16:00:04 installation.Every time I try to use an iax
channel or register an iax user, I get a Segmentation fault.
Trace:
-- Executing Dial("SIP/25-0368",
"IAX2/25|20|Tt") Segmentation fault
[EMAIL PROTECTED] root]# Ouch ... error
Does anyone know what this mean?
Apr 28 11:53:51 WARNING[907]: chan_iax2.c:6039 socket_read: Received
mini frame before first full voice frame
Thanks,
Daniel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Matthew Boehm wrote:
I've got 5 agents who login/logff via AddQueueMember. Each time they do so,
their stats get reset. Is there anyway to keep these stats across logins?
Nobody has implemented that to date that I am aware of.
___
Asterisk-Users mailing
Hi All.
I have installed Asterisk on linux Redhat version 9, I follow step by ssstep
the installation, my card digium is TDM400P, whith modprobe wcfxs I have
load this module.
My vonfiguration files are in /etc/asterisk, the file /etc/zaptel.conf hace
the folloeing lines:
fxoks=1
#fxsks=4
Anton Krall wrote:
How do you do it? I mean, if a caller is already on the queue and suddenly
all agents logoff.. How do you make the caller fall out of the queue and
into an IVR where he can leave a message?
Have you read the sample queues.conf file? There is an option there
called
Hey Guys,
I lost a deal to another vendor because he could point to a number of
installations of the product he was selling in our area and nationally, even
though _he_ didn't implement them directly. Very frustrating, but I don't
imagine uncommon as we compete with other more recognizable
did you do the following
service zaptel stop
service zaptel start
then run asterisk...
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Victor Alvarez wrote:
Hello,
I can't use IAX with my last CVS-NHEAD-04/28/05-16:00:04 installation. Every
time I try to use an iax channel or register an iax user, I get a Segmentation
fault.
Trace:
-- Executing Dial(SIP/25-0368, IAX2/25|20|Tt)
Segmentation fault
[EMAIL PROTECTED]
You need to change the line type in either your zapata.conf or your
zaptel.conf they need to match.
- Original Message -
From: Luz Lopez [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, April 28, 2005 11:01 AM
Subject: [Asterisk-Users] start asterisk
Hi All.
I have
Hi all,
Does anyone know what the easiest way is to delete voicemail for one
extension? Had a search online but couldn't find anything.
Cheers,
Damian.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Thu, 28 Apr 2005 16:01:44 +
Luz Lopez [EMAIL PROTECTED] wrote:
Hi All.
I have installed Asterisk on linux Redhat version 9, I
follow step by ssstep the installation, my card digium is
TDM400P, whith modprobe wcfxs I have load this module.
My vonfiguration files are in /etc/asterisk, the
I just bought one of these zyxel wireless phones, of course there is no
transfer key. Is there a patch for the stable 1.0.7 that I can implement #
or any other key or combination to initiate a transfer?
I looked briefly through the wiki and archived lists and didn't see much.
Command line on the box and navigate to the directory for your VM.
An example of one of mine...
/var/spool/asterisk/voicemail/default/1003/INBOX/
Issue the rm *.* command
Bye bye files
Your location may vary slightly depending on what * you are using.
I am on AAH 0.9.
Cheers,
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Thursday, April 28, 2005 8:31 AM
To: Adam Goryachev
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Confused on G723 and G729
I'll gladly pay
You would need a transcoding license between the Asterisk PBX and the
G711 phone...
On 4/28/05, Matt [EMAIL PROTECTED] wrote:
So
[g729 provider] -(SIP or IAX)--- [g729 asterisk server]
This is how I'd be setup.. actually more like this:
[g729 provider] --(sip)
Hi List,
I was looking for, but I couldn't find any product or project like BIND
that works with VoIP in an homologous way.
I mean, is there anybody working in a way to register user-ids or domain
name-like information so VoIP calls can be dialed in a number string
format from any IP phone?
I believe that you have to start openh323gatekeeper before starting asterisk.
Regards,
Neal
-Original Message-
From: [EMAIL PROTECTED] [SMTP:[EMAIL PROTECTED]
Sent: Thursday, April 28, 2005 7:05 AM
To: asterisk-users@lists.digium.com
Subject:[Asterisk-Users] asterisk-h.323
Asterisk Users / Asterisk Biz List Members,
About a week ago I cross-posted a message titled Large Asterisk Setup
(~500 Concurrent Calls + Scalability) to Asterisk-Users and
Asterisk-Biz. For reference, the threads generated by that message are
archived at the following locations:
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
It almost sounds like there needs to me a new manager action:
Action: Bridge
ChannelA: SIP/199testfone-1f3c
ChannelB: Zap/6-1
It sounds like the intrinsic functionality for 'bridging' is already there in
Asterisk (duh!), it just needs to be encapsulated in a manager action.
Yes, we need
Take a look to ENUM http://www.enum.org/
- Original Message -
From: Andres Paglayan [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 28, 2005 6:39 PM
Subject: [Asterisk-Users] BIND VoIP anyone?
Hi List,
Hi all
Am looking for a way to generate like 300 simultanious calls to test *'s
perfomance on a big load. * is currently working perfectly with H323,
sip and IAX. Any suggestions are welcome
Sam Njenga
___
Asterisk-Users mailing list
Hi, I have installed zaptel, but I haven't in /etc/rc.d/init.d the file to
start zaptel.
From: Jerry Geis [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] start
Folks, I'm having trouble with my voicepulse numbers.
Over the past week, incoming calls have been very
slow to be answered, but they seem fine while the call
is in progress. When the caller hangs up, asterisk
takes a while (over 2 minutes in some cases). This
system does not make outgoing
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
I've been there.. the page comes up with There are currently no files
for this type.
Well, you either have a technical problem or an administrative one.
Eliminate the possibility of corrupted cookies or browser cache by
going to another workstation, accessing
Don't forget Dundi is such a system that is already integrated into
Asterisk.
http://www.dundi.info/
Original Message
Subject: [Asterisk-Users] BIND VoIP anyone?
From: Andres Paglayan [EMAIL PROTECTED]
Date: Thu, April 28, 2005 11:39 am
To: Asterisk Users Mailing List -
Henry Devito wrote:
I just bought one of these zyxel wireless phones, of course there is no
transfer key. Is there a patch for the stable 1.0.7 that I can
implement # or any other key or combination to initiate a transfer?
I looked briefly through the wiki and archived lists and didn't see
Has anyone come across any software that can control adding/editing
SIP extension properties and perhaps dial plan properties on a context
basis. What I mean is I would like it so an admin user from Company A
can manipulate
properties for extensions in his context but not in another
1 - 100 of 214 matches
Mail list logo
