On Tue, Jul 31, 2007 at 10:04:42PM -0400, hugolivude wrote:
Hi,
I'm having trouble compiling zaptel 1.4.4 on SUSE 10.1. I'm really
only interested in getting ztdummy to work because this is a dev
machine with no zaptel h/w. SUSE 10.1 is a 2.6 kernel:
asterisk-dev:/home/hugh # uname -r
On Tue, 31 Jul 2007, Jeng Yu wrote:
Hi All,
I have a telephony project for which I need
to build a prototype to demo for management.
The prototype must work on a GSM phone network.
In the demo system, a call from GSM phone comes
into the demo box. The demo box runs CallWeaver.
What kind of switch are you connecting the phones to? I've seen that
behaviour with cheap Repotec switches (24+2Gigabit). Just replacing it
with a different one fixed the problem.
Julian J. M.
On 7/31/07, Tom Lanyon [EMAIL PROTECTED] wrote:
The issues:
Dropouts - by far the most
Am Dienstag, den 31.07.2007, 07:39 -0500 schrieb Asterisk guy:
1and1 dedicated server's service has been down for a few hours ,
unable to reach them by phone or email. do anyone know what is going
on there ?
There were rumours they had trouble with an outdated version of the
web
Hi Andrew
Thanks for your response.
Yes, the option is progressinband in sip.conf
As far as I can tell, the progressinband setting does not prevent the
183 from being sent altogether. For example, with
progressinband=never, early audio from UA1 to UA2 after a 180 ringing
will still trigger a
Right. autoconf.h is not necessarily generated. Please ignore that
warning. To be fixed in upcoming versions of Zaptel.
(This is not related in any way to GNU autoconf, that is used to
generate the ./configure script. The program autoconf itself is not
needed to build Zaptel, unless you
./configure
Works
make menuselect
Works
make
make install
Does not work. Errors already posted.
menuselect select the modules to compile
do you have problems with this?
I can select modules but make fails.
Thanks,
Hugh
___
--Bandwidth
Hello good ppl,
A couple of questions for multiple pbxes
1. Is it possible to support multiple pbxes in one Asterisk box(using
contexts, etc.)?
2. Can we use the domain field in sip.conf to specify the different
domains for sip users, having one domain for each pbx?
I just tried registering two
On 7/31/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
span=1,1,0,ccs,hdb3,crc4
I was under the impression that ccs/hdb3 was more typical of E1 service than
T1.
I ran across this when looking up something on span syntax yesterday (from
On Wed, Aug 01, 2007 at 06:41:42AM -0400, hugolivude wrote:
Right. autoconf.h is not necessarily generated. Please ignore that
warning. To be fixed in upcoming versions of Zaptel.
(This is not related in any way to GNU autoconf, that is used to
generate the ./configure script. The
On Wed, 2007-08-01 at 11:44 +1000, Klaverstyn, David C wrote:
I’m having problems trying to get a TE120P operational in Canada.
[snip]
I’m not sure if the span line is correct.
[snip]
span=1,1,0,ccs,hdb3,crc4
This would be right if you're configuring an E1 line, but since you're
in Canada
Hi all,
This is the first time i am using stun with asterisk for nat problems. I
have read the rfc which describes how stun works. i didnt have any problems
understanding it. I have also intalled the stun server called stund which i
downloaded from sourceforge. I have seen on the list that most
STUN is a pretty simplistic server. There's nothing else that needs to
be configured on the STUN server side. It's pretty much either running
or it's not.
Just start plugging in the server to your clients and give it a whirl.
It should work.
N.
Rizwan Hisham wrote:
Hi all,
This is the
Guys,
please help me in understanding what I'm mistaking...
Description:
I've configured my AsteriskNOW (beta 6) server, in service providers
section, with the parameters provided by my ITSP. Until now I've used
this configuration with SJphone and all worked perfectly.
Now I've decided
I cannot really help except to say you may want to ask this question on
the stund list (if they have one) since it relates more to the STUN
software than it does Asterisk.
Thanks,
Steve
Rizwan Hisham wrote:
Hi all,
This is the first time i am using stun with asterisk for nat problems.
I
Hello,
I am a small business owner in need for a solution
that automatically answers an incoming call, prompts
the caller via touch-tone menu (press 1 to leave a
message, press 0 to speak to a representative) and
will ring my (real) phone ONLY if requested by caller.
I know that Asterisk is
Can you please get rid of your awfull long nonsense disclaimer?
On 8/1/07, Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello good ppl,
A couple of questions for multiple pbxes
1. Is it possible to support multiple pbxes in one Asterisk box(using
contexts, etc.)?
2. Can we use the domain field in
Does anyone have any tricks to use some logic with SIP UA's codec
negotiation based on the UA's IP? What I would like to do is have Cisco
7960's use g711u when they register with a local IP, and g729 when they
register with a non-local IP. I was thinking about sip.conf and making two
entries for
Lynn,
If I understand you question correctly, you would need:
A computer (preferably a server) to run Asterisk
An analog interface card such as the Digium TDM400P
An analog phone line (POTS)
An analog (real) phone
Calls would come in on the POTS line, get answered by Asterisk. Callers would
Lynn,
What you need is an ATA (analog telephone adapter). The ATA is a SIP or IAX
extension on your Asterisk box, and your standard phone plugs into it.
Asterisk sends the call to the SIP extension (the ATA), and the ATA rings
your phone. On the flip side, your phone dials normally and the ATA
Quoting Linux Lover [EMAIL PROTECTED]:
any of the various module based cards with one fxo and one fxs port
will do what you want.
Hello,
I am a small business owner in need for a solution
that automatically answers an incoming call, prompts
the caller via touch-tone menu (press 1 to
On Wed, 2007-08-01 at 06:48 -0700, Linux Lover wrote:
I am a small business owner in need for a solution
that automatically answers an incoming call, prompts
the caller via touch-tone menu (press 1 to leave a
message, press 0 to speak to a representative) and
will ring my (real) phone ONLY if
Ok thanx. One more thing to ask is: does asterisk has a stun server
implemented in it or not. i mean does asterisk contain a stun server and
provides any application which can be used for enabling the stun server in
asterisk?
On 8/1/07, SIP [EMAIL PROTECTED] wrote:
STUN is a pretty simplistic
No... there's no STUN server built into Asterisk. Asterisk handles NAT
in a different way... and is an endpoint rather than a proxy, so it
doesn't really NEED STUN built into it.
However, we run a STUN server on the same machine as an Asterisk server
and see nothing in terms of load increase.
Yes, you understood correctly. Thank you - and all
others who replied so quickly - for your precise and
guiding answers.
The Digium TDM11B looks looks like the perfect match
for me:
http://www.telephonyware.com/telephonyware/tw00068.html
But one thing that I forgot to mention is that my
Hi, All,
I have a question about agents and queues. Right now we have about 4
queues in our system. Some agents are in multiple queues. Our main
queue is for technical support and it's by far our busiest queue as
well. We have the autologoff feature set to 14 sec right now in the
agents.conf
which stun server do you use?
On 8/1/07, SIP [EMAIL PROTECTED] wrote:
No... there's no STUN server built into Asterisk. Asterisk handles NAT
in a different way... and is an endpoint rather than a proxy, so it
doesn't really NEED STUN built into it.
However, we run a STUN server on the same
Hi Lynn,
You can use a Linksys SPA-3102 for both FXO (POTS) and FXS (phone)
connection instead of a Digium card. The price is around $90-100.
Almost any old PC will do if it can run Linux. There are also other
alternatives to a PC such as the Linksys WRT54GL. I use a Linksys NSLU2
(Slug) at
Hi,
I have an Asterisk 1.2 (can`t upgrade to 1.4 because of some makefile error
on my particular system, bug report opened). That being said, I doubt my
particular issue is a bug, I think it's me not understanding something.
Let`s take a simple dialplan command, i.e. make the phone ring for
Lynn,
I am unfamiliar with soho-pbx, so I cannot comment on quality, service,
configurability, etc. They are based out of Hong Kong, and their box is
probably already running some flavor of Asterisk, so you would need nothing
additional except for the phone line coming in and the telephone.
This is what I have at home and it works okay. I also added an SPA-2002
(~$70) that adds another two FXS (phone) ports for a total of three.
Godspeed,
Phil
Drew Gibson wrote:
Hi Lynn,
You can use a Linksys SPA-3102 for both FXO (POTS) and FXS (phone)
connection instead of a Digium
On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote:
But one thing that I forgot to mention is that my
business is only in its beginning stage and I need to
be as thrifty as possible. While $216 is a reasonable
price, I was wondering whether my (currently very
modest) goal can be achieved by
A phone system for under $100 is asking a lot.
It can be done, but what is your time worth.
You might want to consider some other phone system if all you need is
IVR and analog support or look at hosted solutions.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
Do you think you'll outgrow 1 phone line any time soon. If so You'll
want something that you don't have to completely redo when you add the
next line. That digium card you linked to has 2 more expansion slots
open for more lines or phones.
The soho pbx you linked to looks like you can have
I've been working with a telco for the past two days trying to get a
PRI span up and running. This is a small-ish telco and I get the
feeling they don't do this very often. Anyway, they specified a
pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc.
All of my b-channels are
Linux Lover wrote:
But one thing that I forgot to mention is that my
business is only in its beginning stage and I need to
be as thrifty as possible. While $216 is a reasonable
price, I was wondering whether my (currently very
modest) goal can be achieved by spending much less
(under
Hello All,
I apologize for the slightly off-topic question, but I'm sure that the
people best acquainted with the issue would be hanging around here.
We recently deployed several Linksys POE switches for some smaller customers
(10-24 station) and appear to be suffering from intermittent lock-ups
Hi All,
I remember some folks had put together a web page or perl script to
generate many sip.conf entries from a file defining the users, vmbox,
secret, CID and other variables.
Can someone please point me in the right direction.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
On 7/31/07, Jerry Geis [EMAIL PROTECTED] wrote:
I am trying to re-create calling sendDTMF in an agi and not hearing the
digit either. The above seems to re-create that without the AGI.
...you will have to configure your polycom / sip peer for inband
DTMF if you want to hear the tones.
--
On Wed, 2007-08-01 at 11:43 -0400, Mike wrote:
Aug 1 11:47:57 NOTICE[26107]: app_dial.c:1069 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
This happens when Asterisk don't know where to find the peer (which is
often the case if the device has failed
I use on a regular basis the D-Link line, they work. With the SNOM
you will want to set the ignore Ethernet unplug in case the Ethernet
switch restarts (like a Netgear 7248 attached to a cheap fiber trans).
Keep in mind that holding a GSM phone real close to some of the SNOM
phones will cause
Asterisk Users,
I am running Asterisk 1.2.13 on Debian Linux 2.6.18-4-amd64 and having
problems with DTMF Tones. I have sip service from Teliax and configure to
use rfc2833 for dtmfmode. The problem occurs, when I am using Linksys PAP2T
phone adapter with a regular analog phone.
Is this
what channel are they putting the Dchannel on?
Post your zapata.conf and zaptel.conf
On 8/1/07, Erik Anderson [EMAIL PROTECTED] wrote:
I've been working with a telco for the past two days trying to get a
PRI span up and running. This is a small-ish telco and I get the
feeling they don't do
On 8/1/07, C F [EMAIL PROTECTED] wrote:
what channel are they putting the Dchannel on?
Post your zapata.conf and zaptel.conf
The D channel is on 24.
zaptel.conf:
loadzone=us
defaultzone=us
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
zapata.conf
lpdlnx04 asterisk # cat zapata.conf
;autogenerated
On 8/1/07, John covici [EMAIL PROTECTED] wrote:
I had some troubles -- try setting the timing parameter to 0 (second
one in your span) and see if that helps.
If I'm reading the docs correctly, this param should only be set to 0
if you *never* want to use the T1 connected to this port for
asterisk-dev:/ # rpm -qa | grep kernel
kernel-default-2.6.16.13-4
Thanks,
Howard
On 8/1/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Aug 01, 2007 at 06:41:42AM -0400, hugolivude wrote:
Right. autoconf.h is not necessarily generated. Please ignore that
warning. To be fixed in
I am looking for a retail DID provider which should provide unlimited
incoming calls something around 4-5 bucks . Nufone seemed like a good choice
at $5 but they are down again :( . Any suggestions please .
___
--Bandwidth and Colocation Provided by
John,
On Wed, 1 Aug 2007, John Meksavan wrote:
I am running Asterisk 1.2.13 on Debian Linux 2.6.18-4-amd64 and having
problems with DTMF Tones. I have sip service from Teliax and configure to
use rfc2833 for dtmfmode. The problem occurs, when I am using Linksys PAP2T
phone adapter with a
I had some troubles -- try setting the timing parameter to 0 (second
one in your span) and see if that helps.
on Wednesday 08/01/2007 Erik Anderson([EMAIL PROTECTED]) wrote
On 8/1/07, C F [EMAIL PROTECTED] wrote:
what channel are they putting the Dchannel on?
Post your zapata.conf and
IdeaSIP, Voxbone, Gizmo Project, etc... if we're talking retail.
N.
Mail list wrote:
I am looking for a retail DID provider which should provide unlimited
incoming calls something around 4-5 bucks . Nufone seemed like a good
choice at $5 but they are down again :( . Any suggestions please .
James FitzGibbon wrote:
Another day, another apparant unexplained hardware incompatibility.
I have a TE412P and a TDM400B living quite happily in a whitebox using an
Intel motherboard:
http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm
I tried to move to an IBM x3650
Call Sangoma
On 8/1/07, Erik Anderson [EMAIL PROTECTED] wrote:
On 8/1/07, John covici [EMAIL PROTECTED] wrote:
I had some troubles -- try setting the timing parameter to 0 (second
one in your span) and see if that helps.
If I'm reading the docs correctly, this param should only be set to 0
I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2
digits into the dialplan.
error
-- Invalid extension '81' in context 'impact' on
SIP/207.174.111.34-b77167f8
I pressed 8107
and ideas
my dial plan is (part of it)
[impact]
exten=s,1,Answer()
Looking at your dialplan I don't see extension 8 anything (8XXX) -- Are
you sure you didn't have those extensions in another context that you forgot
to include?
According to the dialplan it is catching the invalid extension and should be
passing it to the i (invalid) handler to loop back into
Just had to install the linux kernel source...
All better now! Thanks for responding everyone!
I have a few installation questions, but I'll post them in a separate thread.
Hugh
On 8/1/07, hugolivude [EMAIL PROTECTED] wrote:
asterisk-dev:/ # rpm -qa | grep kernel
kernel-default-2.6.16.13-4
Thanks Jared. It answers most of my question. Now, when the device doesn't
register, the behavior is as expected. But eventually, any device that
registers successfully might be unplugged, leaving Asterisk to wonder where
the device has gone.
So, what's the best approach to this? Should I put
I do have it in the inside context. It is also doing the circle dance.
I just gave an example. It seems as if it is just forgetting any digits
over 2. like that is in the dialplan but it is not.
Jason
Anthony Cennami wrote:
Looking at your dialplan I don't see extension 8 anything (8XXX)
Hi,
I'm installing * 1.4.9 and a couple questions have come up:
1) I read
herehttp://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x(
http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x)
that if you are using E1 cards you need to install LIBPRI. I'm not
John, thank you very much. Indeed, this is the
direction I was thinking of taking. I just needed a
quick dirty solution for the short term - I didn't
realize that Asterisk is so complex.
In fact, I am not sure I completely understand it:
Will using Asterisk force me to use an external VoIP
The first release of Voiceglue is now available.
Voiceglue provides a VXML interpreter using Asterisk
telephony and the OpenVXI VXML parsing suite.
It is released under the GPL, and thus compatible
with Asterisk and OpenVXI licensing.
The first release is available at the project website:
Just got one of these. Horrible to program.
Trying to key in the FTP server. Won't even
remember the info after rebooting.
Anybody know the proper way to beat on this
stupid beast so it will work?
___
--Bandwidth and Colocation Provided by
I have a pri connection to the phone company.
Sending DTMF out over the pri I hear on my phone when I call it.
However, a second box uses a SIP connection to talk to the first box.
When the second box is trying to do the function sendDTMF(1) over the
SIP connection and then out the PRI I do not
Don't know about the 320, but we provisioned the 301's. They're
provisioning is basically the same as the 501's and 601's. What problems
are you having?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Doug
Sent: Wednesday, August 01,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of SIP
Sent: Wednesday, August 01, 2007 1:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Retail DID provider ?
IdeaSIP, Voxbone, Gizmo
Quoting Linux Lover [EMAIL PROTECTED]:
John, thank you very much. Indeed, this is the
direction I was thinking of taking. I just needed a
quick dirty solution for the short term - I didn't
realize that Asterisk is so complex.
In fact, I am not sure I completely understand it:
Will using
Well, I don't see it in the [inside] context that you include, and the
Background application, by design, will jump to the invalid context as soon
as it can no longer match a valid extension. I am assuming you're dialing
from [impact,s,3]?
Can you include the entire context(s), since the email
qualify=yes in the sip.conf context for that device will change the device
to unreachable and should send you directly to voicemail. There could still
be a brief period where the device is timed out and the system hasn't
qualified it yet, but outside of that, it will just continue trying to send
On 8/1/07, hugolivude wrote:
Hi,
I'm installing * 1.4.9 and a couple questions have come up:
1) I read here (
http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x)
that if you are using E1 cards you need to install LIBPRI. I'm not using
any cards on this system, so
On Thu, 2007-08-02 at 01:12 +0530, Mail list wrote:
I am looking for a retail DID provider which should provide unlimited
incoming calls something around 4-5 bucks . Nufone seemed like a good
choice at $5 but they are down again :( . Any suggestions please .
I'm using www.les.net .
Regards,
On Wednesday August 01 2007 5:49 pm, Douglas Garstang wrote:
Don't know about the 320, but we provisioned the 301's. They're
provisioning is basically the same as the 501's and 601's. What problems
are you having?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
On Wed, Aug 01, 2007 at 05:31:59PM -0400, hugolivude wrote:
Hi,
I'm installing * 1.4.9 and a couple questions have come up:
1) I read
herehttp://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x(
http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x)
Thanks for the help it was a provider issue
Jason
Anthony Cennami wrote:
Looking at your dialplan I don't see extension 8 anything (8XXX) --
Are you sure you didn't have those extensions in another context that
you forgot to include?
According to the dialplan it is catching the invalid
Alex,
The DTMF tones are being sent twice. On SIP Peer side, I set the
DTMFMODE=RFC2833 and the PAP2T you can choose from INBAND, AVT, INFO, and
AUTO, so I chose Auto. Should change on Peer Side and the PAP2T side to use
INBAND?
From: Alex Balashov [EMAIL PROTECTED]
Reply-To: Asterisk
The DTMF tones are being sent twice. On SIP Peer side, I set the
DTMFMODE=RFC2833 and the PAP2T you can choose from INBAND, AVT, INFO, and
AUTO, so I chose Auto. Should change on Peer Side and the PAP2T side to use
INBAND?
From: Alex Balashov [EMAIL PROTECTED]
Reply-To: Asterisk Users
Thanks. Tell me, how intensive is it to use qualify? What type of
packet/check is done with this? Is it reasonnable to use qualify for
thousands of devices?
Once the device is considered to be unreachable for any number of reasons,
will another poll of the device be done to check if it became
Once again guys thanks so much!
Baji - Took a look at your instruction page. Thanks for putting that
together. I've bookmarked it!!
Howard
On 8/1/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Aug 01, 2007 at 05:31:59PM -0400, hugolivude wrote:
Hi,
I'm installing * 1.4.9 and a
John Meksavan wrote:
The DTMF tones are being sent twice. On SIP Peer side, I set the
DTMFMODE=RFC2833 and the PAP2T you can choose from INBAND, AVT, INFO, and
AUTO, so I chose Auto. Should change on Peer Side and the PAP2T side
to use
INBAND?
From: Alex Balashov [EMAIL PROTECTED]
Mike wrote:
Thanks. Tell me, how intensive is it to use qualify? What type of
packet/check is done with this? Is it reasonnable to use qualify for
thousands of devices?
Once the device is considered to be unreachable for any number of
reasons, will another poll of the device be done
James, thank you for your educating answer.
--- James FitzGibbon [EMAIL PROTECTED]
wrote:
This SOHO PBX box won't interop with Asterisk
because it doesn't speak any
of the protocols that Asterisk does. This box
appears to be a solid-state
(and I'd assume very feature restricted)
At 16:49 8/1/2007, Douglas Garstang wrote:
Don't know about the 320, but we provisioned the 301's. They're
provisioning is basically the same as the 501's and 601's. What problems
are you having?
Have no problems with 501s or 601s or 430s.
I punch in the provisioning server IP, but
the phone
At 21:02 8/1/2007, Doug, wrote:
At 16:49 8/1/2007, Douglas Garstang wrote:
Don't know about the 320, but we provisioned the 301's. They're
provisioning is basically the same as the 501's and 601's. What problems
are you having?
Have no problems with 501s or 601s or 430s.
I punch in
Hello all,
I downloaded and built the Asterisk v1.4.9 from the Debian Unstable
repository on my Debian Etch GNU/Linux but when I checked the logs, I got
some error messages from the chan_sip.c. You can find the logs below.
# pwd
/usr/src/debian/
# apt-get build-dep asterisk
# exit
$ cd
you would still need an fxo port of some sort for asterisk for it to
pretend to be a phone.
Quoting Linux Lover [EMAIL PROTECTED]:
James, thank you for your educating answer.
--- James FitzGibbon [EMAIL PROTECTED]
wrote:
This SOHO PBX box won't interop with Asterisk
because it
Ouch.
And I thought I had an answer to my query.
I totaly agree abt the long disclaimer nonsense Schmaltz, but I swear by
the powers up there, it's the admins over here at my workplace doing all
that nonsensical magic, as the mails go out. I wish i had the freedom to
use gmail(just like you),
You can use a Linksys SPA-3102 for both FXO (POTS) and FXS (phone)
connection instead of a Digium card. The price is around $90-100.
Almost any old PC will do if it can run Linux. There are also other
alternatives to a PC such as the Linksys WRT54GL.
The OpenWRT (on whatever supported router
Hi all,
Can I ask that you please keep my personal address in the To: or CC:
in this thread as for some reason I'm only getting half of the list
emails coming through, and they're not showing up on the digium
pipermail archive either. The list archive on http://marc.info seems
to have the
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