I've just migrated Asterisk from my old Gentoo system to an Ubuntu system,
copied across all the /etc/asterisk files and now it fails to work. After
brief looks, I find that it can't access:
/var/log/asterisk/messages
/var/run/asterisk.ctl
/var/run/asterisk.pid
So I touched these files, and
On Wed, 18 May 2005, Steve Underwood wrote:
The header is always in the received image. The TIFF file contains
exactly the same image that a receiving FAX machine would print out.
I think he is refering to the remote fax id to be presented, not the
header. I.e. the 20 digit user selectable
On 5/17/05, Matthew Walster [EMAIL PROTECTED] wrote:
I've just migrated Asterisk from my old Gentoo system to an Ubuntu system,
copied across all the /etc/asterisk files and now it fails to work. After
brief looks, I find that it can't access:
/var/log/asterisk/messages
All:
First let me thank everyone for the good words. It is much appreciated by
all of us at VoIPSupply.com. All of our numbers are up and working. There
are instances from time to time, when T's or PRI go down and we are
without phones services for a few minutes, but this is always kept to a
JD:
Your are correct. B2 Technologies is our parent company.
Thanks,
Garrett
I've ordered several things from them; all arrived as expected.
Last time I ordered from voipsupply but the order was fulfilled by B2
TECHNOLOGIES LLC (same company I think).
JD
Manjit Riat wrote:
I am going
On Wednesday 18 May 2005 07:15, snacktime wrote:
Debian has it's own way of installing asterisk. You should probably
install asterisk again, then copy over only the files you need from
your gentoo box instead of copying the whole directory over.
The only files I've changed are extensions.conf
On Tue, 2005-05-17 at 17:04 +0100, Seb Auriol wrote:
In fact, this is what I'm doing at the moment on the production system, but
we've had a complaint because it doesn't start at the beginning for each
caller. This is pretty important because the file starts with something like
Thank you for
On Wednesday 18 May 2005 07:15, snacktime wrote:
Debian has it's own way of installing asterisk. You should probably
install asterisk again, then copy over only the files you need from
your gentoo box instead of copying the whole directory over.
Oh. Dear God. I just did apt-get remove
Can anyone help me
to understand what the significance of this output is?
May 17 10:50:23
DEBUG[2030]: Didn't get a frame from channel: SIP/105-1ae4May 17 10:50:23
DEBUG[2030]: Bridge stops bridging channels SIP/105-1ae4 and
SIP/outbound-7dc3
I searchedfor
these phrasesbut am coming up
On 17/05/05, Matt Scott [EMAIL PROTECTED] wrote:
Dear all,
I have an asterisk sip issue which I don't believe is unique.
I use a registrar (sipgate.co.uk) where I have 3 different accounts.
These accounts provide me with three seperate local phone numbers which
allow me to allocate them to
I can get you New 7960's for $299.99 each + Shipping or Refurb for
$259.99 each plus shipping.
Can get better prices for qty discount. Which Polycoms are you looking for?
Email me off list
Kyle
[EMAIL PROTECTED]
Manjit Riat wrote:
Looking for 7960s and a few Polycom IP300s and IP600s
Have
Peter Valkov wrote:
John Daragon wrote:
Peter Valkov wrote:
I have build asterisk from latest CVS HEAD-05/09/05 with H323 support
as described in README file.
Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2
kernel-2.6.11
I tested it with following phones: -- XLite
Lastly, we do charge for technical support. We are hear to help, but the
low margins on ATA's etc certainly does not leave us room to give away
free support. All of you that are ITSP's know exactly what I am talking
about.
If you order something, and you can't get it to work, you can pay
B. ffs!
/Danny
On Tue, 2005-05-17 at 22:39 -0500, Brian Capouch wrote:
Chris Mason wrote:
I have gotten
What language is that?
Found in an English dictionary:
get
v. got, (gt) gotten, (gtn)
v. tr.
You don't like the rules?
B.
Is iaxtel down? Im trying to dial Echo test: 1700613 and I get a busy
signal...
Also, is the gw to FWD users down too?
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Hi Guys
I have installed an * system and we seem to have loads of
echo problems. Sometimes worst than others. I have googled and voip-info
ed my little mind out. I am running 3 x zaphfc cards in the machine. Not sharing
irqs, other than themselves. It is on a PIII 1Gig machine with
Hi Gurus.
I searched the lists, wiki and the rest of the web but I still do not
understand this.
My Setup is as follows:
[ISDN via chan_capi or IAX2 DiD Provider] = [* PBX] = [IAX2 Clients
(Atcom AT-320ED)]
I want to get callgroup/pickupgroup and callwaiting working on the IAX
phones. Some web
hi list,
how can i organize several pcs installed with asterisk and e1 cards to be
seen from an asterisk server as one? so if there is a voip call that needs
to be forwarded towards the pstn the asterisk server should find a pc that
has free channel on it's e1 cards that is connected to the
Hi,
Please could any one tell me how could I configure Asterisk inorder to be able to use my modem (instead of FXO cards ...) for outgoing calls. And which type of modems work with Asterisk ?
Do I have to do some changes on Asterisk's scripts or, maybe, add some ones ?!
Thanks in advance.
On Wed, 2005-05-18 at 11:21 +0200, ALIF Mohssine wrote:
Hi,
Please could any one tell me how could I configure Asterisk inorder to
be able to use my modem (instead of FXO cards ...) for outgoing calls.
The simple answer is you can not.
And which type of modems work with Asterisk ?
None
Hi,
I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
The versions i'm running:
kernel-2.4.27
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
zaptel modules 1.0.7
zaphfc is from bristuff-0.2.0-RC8e
When I'm doing the insmod on
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based - no real phone lines).
I tried this (from voip-info.org wiki)...
Hello Dave,
Could I know why please ?? Thanks !Dave Cotton [EMAIL PROTECTED] a écrit:
On Wed, 2005-05-18 at 11:21 +0200, ALIF Mohssine wrote: Hi, Please could any one tell me how could I configure Asterisk inorder to be able to use my modem (instead of FXO cards ...) for outgoing calls.The simple
I should mention that I have tried using the call forward function of
the sip phones, but a) this means configuring the phones and some are
remote and behind firewalls and b) It doesn't work...
Cheers,
Mark
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Dear Nicolas Olivier
Just try the florz patch at http://zaphfc.florz.dyndns.org/
and look at cat /proc/interupts if your not sharing irq's
Maybe this will help
Good luck
Sjaak
Hi,
I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up
I recently experienced weird buffer overrun errors with zaphfc which I
eventually identified as being was caused by mismatched memory on the
motherboard.
You might want to check this out.
Stuart
Nicolas Olivier wrote:
Hi,
I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is
Just an update, I deoopsed the kernel dump, must be usable...
Nicolas Olivier wrote:
Hi,
I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
The versions i'm running:
kernel-2.4.27
Asterisk
Good day all
Did anyone get the eicon 4 bri working with asterisk and fedora core 3
Please
Thanks
Altus
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Well, afer applying zaphfc_0.2.0-RC8a_florz-6.diff, I'm highly flooded after
ztcfg with:
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer underrun:
0, 0
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer overflow:
311, 311
May 18 18:11:33 gw-ss
Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using
Arrggh Nuts.
Don't suppose anyone has a spare NM-HDV hanging around that they want to
sell ?
:(
Julian.
[EMAIL PROTECTED] wrote:
You need an NM-HDV card of some sort to run voice. The WIC-1MFT-E1
can handle voice, but you still need the DSP's to use it as a voice
card. Putting that into an
In 18/05/05, Mark Benson [EMAIL PROTECTED] wrote:
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based - no real phone
This afternoon we were discussing, and found that we would like one box,
which should have ALL of these:
1. WAN port
2. Ethernet port 1 with Power over Ethernet
3. Ethernet port 2 with or without PoE
4. FXS port
5. FXO port
6. DHCP, web configureable.
7. Optional wireless accesspoint
8. One and
Hi all,
as in last mail, i've installed Asterisk from CVS and AMP to manage it. I've made 4 extensions:
moloch*CLI sip show peers
Name/usernameHostDyn Nat ACL Mask Port Status
204/204 (Unspecified)D 255.255.255.255 0UNKNOWN
203/203
Hi
I have installed 2 x Eicon BRI ISDN cards into a Suse 9.2 server. We can
use all 4 lines for out going calls fine, but on incoming we can only use 2.
On calling in using the main msn, the 3rd line gives a an engaged signal.
I have unplugged 1 of the cards, and the other card takes the 2
[EMAIL PROTECTED] wrote:
Hi
tks for the feedback, the admintool i cant use, because users
create/add themselves to the system themselves, could be 100 or 1000+
users. Hence I could get my script which create user/pass details in
myqsql to call the voicemail script to create the physical path
doesnt invetel do one
Iqbal
Ronald Wiplinger wrote:
This afternoon we were discussing, and found that we would like one
box, which should have ALL of these:
1. WAN port
2. Ethernet port 1 with Power over Ethernet
3. Ethernet port 2 with or without PoE
4. FXS port
5. FXO port
6. DHCP, web
I asked my friend to setup FWD and call me to my *
However, it did not matter which codec we used, after three seconds the
connection was cut.
Why? and how to make it stabled?
bye
Ronald
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Er... set the trunk variable to what? I thought it was a built in
variable...
Peter Bowyer wrote:
Have you set the TRUNK variable in the [globals] section of
extensions.conf? Looks like you didn't.
Peter
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I've just give a look to the website
http://www.voip-info.org/wiki-Asterisk+Hardware
If I understand very well, the Intel modems marked with 537 or MD3200 chipset should work with Asterisk ?!
If it is true, I'd like to know how to configure Asterisk ?
Thanks a lot.
Découvrez le nouveau Yahoo!
Hi all,
I am having a problem with a recent installed *. The IVR,
voicemail internal greeting sounds dont play!. I see on the CLI
interface that it is playing but I cant hear anything.
I have the following configuration on the asterisk.
-
Current Asterisk CVS
-
A TDM400 with 4
On 18/05/05, Mark Benson [EMAIL PROTECTED] wrote:
Er... set the trunk variable to what? I thought it was a built in
variable...
No, it's not. Looking at your dialplan extract, you need to set TRUNK
to the name of the trunk to place the outgoing call on.
eg
TRUNK=IAX/voiptalk
You might need
AVE!
i am trying to register h323 asterisk to the gatekeeper as i installed
asterisk, libpri, zaptel from CVS, and pwlib, openh323, asterisk-oh323
on fedora core3 on a cisco mcs 7800 server problem is i want the
asterisk to register with gatekeeper endpoint with specific zone name
and type...
i
I have been able to get it working by explicitly setting the dial command...
So should the trunk variable be the divice to dial out on?
Mark Benson wrote:
Er... set the trunk variable to what? I thought it was a built in
variable...
Peter Bowyer wrote:
Have you set the TRUNK variable in the
Hi Peter.
I think I probably put my email rather badly.
However you did manage to spot my problem and solve it for which I am very
grateful!!
The bottom line is you cannot have different context for the same sip
provider, and it works as you state in your reply.
Thanks again.
Matt
-
Being around the internet for a quite a long time this gives me an
uneasy feeling. I have seen companys start to go under and pull the
plug when they get into financial trouble(not being able to pay the
bills) and run with the customers money. I have had this happen to me
on 2
Nicolas,
I replied earlier stating that I saw similar issues and now that you
have applied the Florz patch the symptoms you are seeing are all but
identical to the issues I saw and resolved by changing out the
motherboard memory. The system was an ASUS main board with a Xeon processor.
It is
HelloOn 18/05/2005, at 4:09 PM, Peter Svensson wrote:I think he is refering to the remote fax id to be presented, not the header. I.e. the 20 digit user selectable number on the remote fax. The one often seen on the lcd of the receiving fax and so on. Yes that's exactly what I'm referring
Hi, I´m trying to migrate my propietary software to an asterisk server
connected to a Ericsson BP 128i PBX.
I´ve been looking at the asterisk web, user forums, published docs about how to
use the PBX as the hardware device but I haven´t found anything.
I think this is possible. The old server
On May 18, 2005 06:45 am, Iqbal wrote:
doesnt invetel do one
Got a link? Googling for invetel comes up with car counters and stuff...
nothing really VOIP related.
-A.
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El 18/05/2005, a las 11:42, Mark Benson escribió:
-- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1,
/07961106nnn|20|r) in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable
Thanks,
Staring to see where I was going wrong. Now I know the explicit dial
string (as you say I tried that in the dial plan and it worked) I can
mess around with the trunk variable.
Cheers!
Peter Bowyer wrote:
On 18/05/05, Mark Benson [EMAIL PROTECTED] wrote:
Er... set the trunk variable
I'm having some troubles with my * machine, when i place a call on hold
the callee doesn't hear any MOH and the call is dropped because of lack
of RTP.
I also don't see * starting MOH on the SIP channel the callee is on (moh
class is defined, there are MP3 files and mpg123 is active).
I'm using *
On May 18, 2005 07:22 am, Jean-Yves Avenard wrote:
Yes that's exactly what I'm referring to.
Most fax machines I've used print this information on the top left
corner or top right corner on any fax received.
Is it possible to do this with SpanDSP?
You can get the info and stamp it into the
Nicolas Olivier wrote:
I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
The versions i'm running:
kernel-2.4.27
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
zaptel modules 1.0.7
zaphfc is from bristuff-0.2.0-RC8e
When I'm doing
On Wed, 18 May 2005, Lee Norvall wrote:
Hi
I have installed 2 x Eicon BRI ISDN cards into a Suse 9.2 server. We can
use all 4 lines for out going calls fine, but on incoming we can only use 2.
On calling in using the main msn, the 3rd line gives a an engaged signal.
I have unplugged 1
HiOn 18/05/2005, at 9:35 PM, Andrew Kohlsmith wrote:You can get the info and stamp it into the image yourself with some third party TIFF manipulation tools, I bet. I wouldn't mind doing so if I knew where this Fax ID information is stored or how to retrieve it, or if it's even possible.JY ---
Wow looks perfect - this will be unreal if this works.
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Peter Valkov
Sent: Wednesday, 18 May 2005 12:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Please could any one tell me how could I configure Asterisk inorder to
be able to use my modem (instead of FXO cards ...) for outgoing calls.
The simple answer is you can not.
And which type of modems work with Asterisk ?
None
Do I have to do some changes on Asterisk's scripts
Marty Mastera wrote:
Can anyone help me to understand what the significance of this output is?
May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel: SIP/105-1ae4
May 17 10:50:23 DEBUG[2030]: Bridge stops bridging channels
SIP/105-1ae4 and SIP/outbound-7dc3
I searched for these phrases
Jean-Yves Avenard wrote:
Hello
On 18/05/2005, at 4:09 PM, Peter Svensson wrote:
I think he is refering to the remote fax id to be presented, not the
header. I.e. the 20 digit user selectable number on the remote fax. The
one often seen on the lcd of the receiving fax and so on.
Yes that's
Hi,
Is it possible to put some kind of bridge which will do traffic
shaping/prioritising between
my 6 external IP addresses and my PPPoA modem interface?
My other option is to put some kind of device at the edge of all my
networks to shape the
traffic in/out. I'd rather do it in one box if
Hi
I can see what seems to be both devices in use, so I guess it must be
down to the capi.conf (below), does this look correct ???
[interfaces]
msn=292880
incomingmsn=292880, 292881, 292882, 292883, 292884, 292885, 292886,
292887, 292888, 292889
outgoingmsn=292880
controller=1
softdtmf=1
[EMAIL PROTECTED] wrote:
Marty Mastera wrote:
May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel:
SIP/105-1ae4 May 17 10:50:23 DEBUG[2030]: Bridge stops bridging
channels SIP/105-1ae4 and SIP/outbound-7dc3
I am noticing these in my logs also. I looks like it is the result
of the
Quoting from:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Zaptel%20Installation
As I haven't got a Digium card, I need a timer which can be provided by
ztdummy, zaprtc or zaprai.
But anyway the results are the same with or without zaprtc loaded.
Peer Oliver Schmidt wrote:
Nicolas
Hi.
We are trying to set up asterisk to service a wireless community in our
town.
We have about 30/40 wireless working nodes each one with a 10.34.x.x/24
subnet for users. Each one of these addresses can potentially have a
192.168.x.x/x subnet.
On top, the wireless nodes, themselves, are linked
Stuart,
I switched the system to a pentium based host, with different memory.
The results are the same. I've also changed the ISDN card to be sure.
Nicolas
Stuart Hirst wrote:
Nicolas,
I replied earlier stating that I saw similar issues and now that you
have applied the Florz patch the
Can anyone explain the Polycom
Text Messaging features built in to the IP 500/600? Can Asterisk (or something else) talk to
it? Ive seen vague references to
MSN Messenger, and somehow thats mentally disturbing
Chris Coulthurst
[EMAIL PROTECTED]
. Snip
It is sad to hear that you will not be purchasing from us. I do not
understand though, why we owe you an explanation for our toll free number
being down.
^^
You are right you don't owe any explanation at all for your numbers being
down. It was your Toll Free and
hello
sip.conf
bindport=5070
i am trying to register at ser 5060. but why i am
getting request at asterisk 5070.
thanks
Kamran
Yahoo! Mail
Stay connected, organized, and protected. Take the tour:
http://tour.mail.yahoo.com/mailtour.html
Hi guys,
I am trying to install Debian sarge (latest netinstall) on ML110 server
with two SATA hardware mirrored drives on Adaptec 2610SA controller for
use with Asterisk with no luck.
Debian installer does not see the array. Any workarounds?
Please help.
Regards,
Alex.
For example, how does your dialplan look on the zap and sip servers in
order
to route the call from a zap on server 1 to a sip on server 2?
If you want any SIP server/client to be able to call you at
[EMAIL PROTECTED], for example, then in the context that is set in the
[general] part of
Pizco,
SER is definitely better suited to deal with NAT issues then ASTERISK is. I
suggest looking at SER and NAT helpers like media proxy application (part of
SER). I also recommend looking at NAT devices at SER wiki page to make sure
that your router/nat device is compatible. In general, this
Sounds like reinvite troubles. Once the SIP endpoints are both in the
call the server (FWD) will get out of the way allowing the two SIP
clients to connect directly. There can be cases where you can connect
through the server but not directly, usually because of NAT traversal
failure at one end or
Mark Johnson wrote:
Mark Brown wrote:
Hi Everyone!
Is there any hope for us newbie plebs who want to also get hold of the
updated Cisco firmware?
I need to get a 7910G updated to work on SIP..
Any help on obtaining the updated firmware quickly and painlessly
would be great J
Cheers
M
7910 does
Hi, I'm using OH323, mostly with success, to interface Asterisk to
a provider's switch (World Telecom INX). I have noticed a particular
effect, and I wonder whether anyone else has seen the same?
The effect is audio flutter (almost like the flutter one gets on
MF or HF radio sometimes) which only
Michele O-Zone Pinassi wrote:
Hi all,
i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones)
and this is what my sip show users return:
moloch*CLI sip show users
Username Secret Accountcode Def.Context ACL NAT
204 moira
I have been successful in setting up asterisk and making workstation to
workstation SIP calls. But I am lost when it comes to anything past that.
We are trying to integrate this asterisk server into with our Executone
(432?) PBX to allow us to make outbound SIP calls between our disparate
Ihave Ext 101 configured as the default for incoming calls. Ext 101 also holds all of the incoming voicemails. How do I access the voicemail for ext 101 remotely? I am lookingto be able to call in from the outside and retrieve all of my messages. When I press *97 during the voicemail outgoing
Hi All,
Im having some trouble getting Asterisk to send DTMF via rfc2833. The
scenario is this:
For purposes of testing software, I have two applications communicating
with each other via DTMF. In between the two applications sits an
Asterisk. The applications require that DTMF be sent via
On Wed, 18 May 2005 00:01:53 -0400, Paul wrote:
Manjit Riat wrote:
I am going to buy some IP phones from them but I sent them an email
couple of weeks ago and got no reply. Has anyone ordered anything from
them? Any other places that I can buy from? Sorry if its a wrong post.
Not getting
Jon Gabrielson wrote:
I am trying to get remote extensions to work correctly with
agents. I have ackcall=yes and have agents logged in to
extension 101 using agentcallbacklogin with extension 101 defined as:
exten = 101,1,Dial(Zap/3/18165551234,20,tTA(custom/presspoundtoanswer))
This setup
On Wed, 18 May 2005 11:45:51 +0100, Iqbal wrote:
doesnt invetel do one
Iqbal
Ronald Wiplinger wrote:
This afternoon we were discussing, and found that we would like one
box, which should have ALL of these:
1. WAN port
2. Ethernet port 1 with Power over Ethernet
3. Ethernet port 2 with
mmh I think you asked to the wrong ML,
this is Asterisk, not Debian installer ML.
Cya.
On Wed, 2005-05-18 at 23:00 +1000, Alex wrote:
Hi guys,
I am trying to install Debian sarge (latest netinstall) on ML110 server
with two SATA hardware mirrored drives on Adaptec 2610SA controller
MSN will only work on 1 ISDN2 line and cannot be spread across 2 ISDN2
lines. From your description I assume you have 2 calls up and the 3rd call
fails. This is because you can only have 2 concurrent calls using MSN on
ISDN2. You will find you have a different number range for the second ISDN2
If
Hi,
I am trying to get to the bottom of a warning i am recieving through
the console.
May 18 13:26:29 WARNING[8281]: chan_sip.c:6837 handle_response:
Forbidden - wrong password on authentication for NOTIFY
Calls are still working. I cannot work out what is causing it.
Asterisk - Ingate -
--On Tuesday, May 17, 2005 5:24 PM -0700 Manjit Riat [EMAIL PROTECTED]
wrote:
I am going to buy some IP phones from them but I sent them an email
couple of weeks ago and got no reply. Has anyone ordered anything from
them? Any other places that I can buy from? Sorry if it's a wrong post.
I
Try changing SetCIDNum SetCallerID and use to SetCIDName as under:
Ex:
---
exten = s, 1, SetCallerID(${CALLERIDNUM})
exten = s, 2, SetCIDName(${CALLERIDNAME})
exten = s, 3, Dial(${ARG2}/${ARG1},${RINGSECS})
exten = s, 4, Voicemail(u${ARG1})
exten = s, 5, Hangup
exten = s, 101, Voicemail(b${ARG1})
Nicolas Olivier wrote:
Quoting from:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Zaptel%20Installation
As I haven't got a Digium card, I need a timer which can be provided by
ztdummy, zaprtc or zaprai.
But anyway the results are the same with or without zaprtc loaded.
Irregardless of
Hello all:
I am trying to use the mysql command to retrieve information from a mysql
database.
my example here was formed from using the wiki reference to using the
mysql command.
The problem is with the fetch command.
Here is the macro code:
Mysql(QueryString=SELECT\ ivr-password\ from\ users\
A while back I converted back to static conf files from a database setup.
However I decided to tackle it again.
The problem that I was experiencing, was, there was no stutter tone on my
sipura 2000 or 3000 when there was a voicemail left at either extension
when I was using mysql setup for peers
I am unsure of what you want to achieve. Do you want to interconnect BP
and Asterisk, or replace BP with Asterisk? What is the purpose of
proprietary software you mention? Please give more details.
Niksa
[EMAIL PROTECTED] wrote:
Hi, I´m trying to migrate my propietary software to an asterisk
Has anyone seen a Softphone with the following features:
1) Utilizes Touch Screen
2) Has API for interfacing CID info with existing application on same PC.
Thanks
Bill Ford
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Hello Rod, I'll try it, thanks.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Wednesday, May 18, 2005 1:01 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Static on TDM Zaptel FXO
Make sure you have disabled
Hi PeterOn 18/05/2005, at 10:05 PM, Steve Underwood wrote:It is only there because the sending machine put it there in the image. Spandsp is not different from how any FAX machine I have ever used behaves. As well as sending the 20 digit number as text, the sending machine puts in the header. This
Hi,
What would you say that the best compression format is for voice
recordings on Asterisk? The tradeoff being the file's size. I like
GSM because of the small files size but the quality isn't great. What
are people finding as a good setting for GSM?
Thanks,
Michael
Hello Bryce,
Gain settings do seem to have an effect. I am going from a Cisco
7960AsteriskZap TDM CardPOTS
Thanks,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryce
Chidester
Sent: Wednesday, May 18, 2005 1:08 AM
To: Asterisk Users Mailing
Hmm, i can re-produce this problem in a way:
- external call to voip
- voip terminate this call
After this, asterisk produce an sigseg like:
I SEND:DISCONNECT port:1 pid:0 mode:TE addr:51400101
-- l3id:20011 cause:16 dad:72 oad:xyxyxyxyxyxyxyxyxy channel:1 port:1
Ouch ... error while
Correction:
The hardware is a Wildcard T100P (not a TE110P)
Thanks!
-Original Message-
From: Geoff Manning [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 18, 2005 9:07 AM
To: Asterisk Users (E-mail)
Subject: [Asterisk-Users] Integrating Asterisk into our Legacy PBX
--Newb
I
Someone say of configure sipp
with asterisk and asterisk with sipp
I have a lot of problem for
sdp
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