Hello All,
I could use a recommendation if anyone has a
moment. I have the T100P but I have not gotten my service yet. I
want to have at least 12 lines of digital voice with DID. Should I just
seek out a PRI ISDN provider or is there something else I should look for?
I want to keep cost
On Wed, 2005-01-05 at 18:38 +1100, PHP Mechanic wrote:
Have you considered setting up a meetme confrence line for them? :)
analog phone = asterisk/tdm11b = pstn
The meetme option is nice, but it doesn't solve the problem. The TDM11B
only
has one FXO, one FXS. To get the effect the
Hi folks!
Ourcompanyare going to buy an E1 line
with Euro ISDN and 30 lines (channels).
This is how it will be configured:
3 Lines, of the total of 30, is going to be for the
company phones, and share one phonenumber (eg. 555-12340).
1 Line will be dedicated to a specific unique
On Tue, 2005-01-04 at 15:34 +0100, Erik Versaevel wrote:
Mark Elkins wrote:
On Tue, 2005-01-04 at 15:45 +0200, Mark Elkins wrote:
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call -
On Wed, 2005-01-05 at 18:38 +1100, PHP Mechanic wrote:
Have you considered setting up a meetme confrence line for them? :)
analog phone = asterisk/tdm11b = pstn
I have played with it. But the problem I'm having is as follows
exten = _1800.,1,Dial(Zap/4/${EXTEN},20,Tr) ; call some
On Wed, 2005-01-05 at 01:01 -0700, Wiley Siler wrote:
Hello All,
I could use a recommendation if anyone has a moment.
It is preferable to not use HTML in email. Just because a font size
looks good on your monitor doesn't mean it is anywhere close to good
anywhere else. Your choosen font
On Wed, 5 Jan 2005, Altus Snyman wrote:
Good day all
I had a look at the extensions.conf sorting
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
What I'm trying to do is route all my cellphone number threw a channel
and all other calls threw the other 3 channels
On Wed, 2005-01-05 at 19:27 +1100, PHP Mechanic wrote:
On Wed, 2005-01-05 at 18:38 +1100, PHP Mechanic wrote:
Have you considered setting up a meetme confrence line for them? :)
analog phone = asterisk/tdm11b = pstn
I have played with it. But the problem I'm having is as follows
Threeway calling is similar. You can make a small impromptu conference
that way with 2 internal phones and an external or 3 internal phones or
even 1 internal and 2 external calls on separate phone lines. All of
these are mixed inside of asterisk and the PSTN is non the wiser.
Thanks for clearing
Hi,
I'm sorry to bring this up again, but I have been googling forever and
whatever solutions are offered don't work for me.
I am using x-lite (the latest build) and trying to use Speex.
When I do call from the x-lite to another SIP phone or PSTN (through Cisco
gateway) My asterisk fills up
Walter Klomp wrote:
Hi,
I'm sorry to bring this up again, but I have been googling forever and
whatever solutions are offered don't work for me.
I am using x-lite (the latest build) and trying to use Speex.
When I do call from the x-lite to another SIP phone or PSTN (through Cisco
gateway) My
I already did testing with the unit you are talking about and the
status is that it is NOT working.
Search the list on IP600 and asterisk.
In brief:
Their Skinny implementation makes the IP600 wait for an extra signal and
refuses to ring any handset without it. The 'original' Cisco 7940 (which
I will certainly try that. Please also let me know your progress..
On Tue, 4 Jan 2005 22:12:23 +0200 (SAST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On Mon, 3 Jan 2005 [EMAIL PROTECTED] wrote:
Has anyone had success using a TE410P card in an HP-Compaq DL380 G4
server?
Thanks
We have SIP-DECT gateways in Pci:
http://shop.acropolistelecom.net/product_info.php?products_id=30language=en
or PCMCIA cards :
http://shop.acropolistelecom.net/product_info.php?manufacturers_id=11produc
ts_id=29
Regards
Benoit
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL
On Tuesday 04 January 2005 14:06, Peer Oliver Schmidt wrote:
Nils Ohlmeier wrote:
What are the chances to get intercom working with a Snom 190 with a
current firmware? Anyone? Is Snom working on this?
yes, Snom is working on this.
This is very good to hear. Do you have any time frame (2
Thanks, the price looks attractive but what about Asterisk support?
From your website : All standard Windows operating systems are
supported. and we are not using Windoze :)
How many concurrent conversations are supported?
Is there a howto anywhere how this could be used with * ?
By looking at
I'm trying to place a call to asterisk using X-Lite. Asterisk is setup
with some Grandstream phones. I can call from one grandstream extension
to another. When I try to an extension with X-Lite, it comes back with
Status of SIP/2.0 404 Not Found. X-Lite is not registered as asterisk
Hi all,
I am attempting to call from softphone to softphone, I am using X-lite to call
another X-lite.
I get the phones to call each other and finnaly connecting, but cannot hear the
voice at all. Is there any ideas as to why this is happening.
(I don't have sound card in my linux server. I
Hi List!
I installed Asterisk 1.0.3 stable on a RHEL rebuild. Due to problems with
* modules refusing to build I replaced the RHEL kernel with stock 2.6.10.
Asterisk seems to be working but when I dial voicemail I hear nothing.
When I hangup I see a message on the console that the calller did
Hi
but did anyone have ever used a Siemens HiPath PBX with Asterisk?
If you made it, please tell me how...
I read that chan_cornet does exist...
http://lists.digium.com/pipermail/asterisk-users/2004-October/069559.html
Is there any Digium Hardware solution for the Asterisk HiPath connection?
use - on the command line for debugging information, there
should be detailed tracking information provided that will help
On Wed, 5 Jan 2005 12:41:23 +0100 (CET), Remco Barende
[EMAIL PROTECTED] wrote:
Hi List!
I installed Asterisk 1.0.3 stable on a RHEL rebuild. Due to problems
On Wed, 2005-01-05 at 17:02, PHP Mechanic wrote:
Howard Lowndes wrote:
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread the
word?
Yes, I have. I'm using a Voicetronix OpenLine4 card. I did have an
Ronald Wiplinger wrote:
The idea:
If I have a customer, who is registered to my Asterisk box as extension,
than I should not need to ask him for a pin code for ASTCC.
How can I set this up?
Have a look at the astcc.agi file:
# Usage-example:
#
# ;
# ; Card-number and number to dial
Hi,
The HG1500 is a HiPath3000 board and i don't have experience with Asterisk and HiPath3K.
What we have is an Asterisk connected to a Siemens HiPath4000 over a H.323 trunk using oh323 and the HG3550 board. It works fine. But the Siemens HG3550 only supports H.323 V2.0 (so not a lot of features
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread
the
word?
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten =
On Wed, 2005-01-05 at 16:50, James Andrewartha wrote:
Howard Lowndes wrote:
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread the
word?
Yes, I have. I'm using a Voicetronix OpenLine4 card. I did have
Hi
I dont knowif Steffen's chan_cornet is working. I emailed him, but with no
result.
Yesterday I read this article
http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom
It has some solutions... but not yet a direct
Asterisk-HiPath connection.
But doesnt Digium have Asterisk-HiPath
Nils Ohlmeier wrote:
What are the chances to get intercom working with a Snom 190 with a
current firmware? Anyone? Is Snom working on this?
[..]
It is allready fixed. So it should work again in the next firmware release
(which usually results in an availability for the end users in terms of days
On Wed, 5 Jan 2005 07:54:43 +0100, Florian Overkamp wrote:
Hi,
-Original Message-
Have you considered setting up a meetme confrence line for them? :)
analog phone = asterisk/tdm11b = pstn
The meetme option is nice, but it doesn't solve the problem. The TDM11B only
has one FXO, one
On Wed, 2005-01-05 at 23:22, PHP Mechanic wrote:
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread
the
word?
What I need more though is examples of anything that needs to go into
Nils Ohlmeier wrote:
BTW the new snom model (no availability dates yet) will have more
LED's then the existing ones. What are the most important or
interesting features (existing or not implemented yet) for the
Asterisk community for this programmable keys with LED's?
The only missing feature
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten = s,1,NoOp(Caller ID on the PSTN line is ${CALLERID})
M. Tried that, but it didn't deliver ${CALLERID}
Did the caller have callerid enabled by their telco ?
Apologies if the format of the email was troublesome. I am accessing my
email remotely via Outlook Web Access otherwise the format would have
been plain text.
So, I need to learn more about voice T1s? Reeally? That would be why I
am posting to the user group in the first place. To learn more.
how can I configure the sample time period (10ms,20ms etc) for codecs?
kapejod told me on IRC that this could not be achived with
configuration, and that I needed to dig into the source to do this.
Can someone please tell me what asterisk normally uses here?
Should the client setting always
What do you mean with *But doesnt Digium have Asterisk-HiPath solutions?*. If you are meaning a connection with Digium cards so...sorry i've never usedDigium cards. Buti thought (see your first thread *connect Asterisk with Siemens HiPath HG1500*), you are looking for a way to connectAsterisk to
Could you please explain or tell me where it is explained the version
and contents of * that is retrieved with CVS.
I am wondering whether there is a change list or something. If you
tell me here I will update the Wiki ;-)
Thanks
John
___
Hello, I thought that my Digium TDM400P would be the right hardware to
support the zaptel timer, and put the following IAX.CONF entry to test,
(trunk=yes) in the example below
[VHAX]
type=peer
auth=md5
username=whoknows
jitterbuffer=yes
;trunk=yes
secret=terriblesecret
host=4.5.6.7
qualify=1200
Hello Steffen,
hey it sounds very good...!!! but what do you mean with *new hipath version doesn't support H.323 anymore*? What version are you talking about? As far as i know the new version of HiPath4000 V2.0 still supports H.323 (STMI2).Steffen Koepf [EMAIL PROTECTED] wrote:
Hello, I dont know
On Wed, 2005-01-05 at 19:52 +0600, Samudra E. Haque wrote:
Hello, I thought that my Digium TDM400P would be the right hardware to
support the zaptel timer, and put the following IAX.CONF entry to test,
(trunk=yes) in the example below
[snip]
But, it didn't work. So I had to comment it out.
I think you need to subscribe to the context where exten 200 exists.
I'm not sure if it'll work with an arbitrary context. You may also
want to try sending the hint to just one phone. I'm not 100% on the
format for sending the hint to multiple phones.
On Tue, 04 Jan 2005 21:40:00 -0700, Nihal
Hello List
I have a customer with an Option 11 with a E1 Card (MFCR2) to a
Audiocodes Mediant 2000 gateway and a Sip netowrk.
at this moment I have working an Asterisk Box as Voicemail for all
the PBX and working like a Charm. but my customer wants to have an
autoattendant IVR for the option
Hello everybody!
happy new year ;-)
I have a problem with inbound calls on my Diva Server PRI
This is my problem:
2005-01-05 15:38:04 ERROR[31716]: chan_capi.c:1696 pipe_msg: did not find
device for msn = 4132
capi.conf:
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
Hello,
but what do you mean with *new hipath version doesn't
support H.323 anymore*? What version are you talking about?
As far as i know the new version of HiPath4000 V2.0 still
supports H.323 (STMI2).
HiPath 3000 - H.323 Support
HiPath 4000 - NO H.323 Support and nothing else but cornet
hi
some time ago, I asked the list of a good book for learning ISDN and
SS7. I don't need to know how to write a channel driver or something; I
just want to know more about the possibilities and what's really sent
back and forth. I was told the book ISDN and SS7: Architectures for
Digital
The asterisk server doesn't need a sound card. The machines that have
the xlite softphone need to have a sound card.
E.g. you have 2 pc's running xlite + the asterisk server. The asterisk
server doesn't need a sound card but the 2 pc's will. (sounblaster
live solved similar problems i was having)
Roy Sigurd Karlsbakk schrieb:
...
Digital Signaling Networks by Uyless Black (ISBN 0132591936) was a good
choice, but this seems sold out. Does anyone know about another book
Hi,
since it's sold out for a longer time, I sold mine used at
amazon.
Roger.
We all mostly know that * as well as various SIP phones support SMS.
While the final setup is somewhat of a mystery, there are reports of
those lucky souls who have it working. We also know that in order to
send an SMS to a mobile phone, we need to connect to some SMS message
center and get the
After looking at the source I had also tried to increase the buffer size
from 8000 to 16000, but that made other codecs (like lin_to_g729) choke,
and
I still had the problem...
I like speex and would like to use it (as I find ilbc a bit too scratchy)
I am running Asterisk
http://www.thirdlane.com/screenshots.htm
(Asterisk PBX Manager from Thirdlane) looks like a
great program for eye candy configuration of
Asterisk.
However it costs lost of $, and Im currently only an experimenter
so to speak.
Anyone advice of a decent alternative that is similar??
Using the new firmware is there still the issue with needing to patch
chan_sip.c, or does it work out of the box? Do you have details on how it
should be implemented within *?
As of now, the hack still applies. It would be wonderful though if somebody
could implement a command line variable
http://www.thirdlane.com/screenshots.htm (Asterisk PBX Manager from
Thirdlane) looks like a great program for eye candy configuration of
Asterisk.
However it costs lost of $, and I'm currently only an experimenter so to
speak.
Anyone advice of a decent alternative that is similar?? Currently,
Jay Milk wrote:
We all mostly know that * as well as various SIP phones support SMS.
While the final setup is somewhat of a mystery, there are reports of
those lucky souls who have it working. We also know that in order to
send an SMS to a mobile phone, we need to connect to some SMS message
We have just finished installing some Sayson 480i phones (
will post a review soon) and I have one issue. I cannot seem to get the *78 ,
*79 ( and like) functions to work. Are these automatically installed with
Asterisk?
Anything required in the extensions.conf or
sip.conf?
When I dial *78 I
Hi List
I'm having problems starting asterisk with asterisk-oh323-0.6.4.
I'm using this versions:
asterisk-1.0.3
asterisk-oh323-0.6.4
openh323-Janus_patch4 + asterisk-0h323 patch
pwlib-Janus_patch4
At starting time, i've this error message
#
Hello everybody,
Ive been trying to solve a problem for several weeks now but it really
beats me.
There are several hard phones connected to an Innovaphone 3000 VoIP gateway.
On the other side I have a SIP softphone connected to Asterisk. The problem
I have is that on incoming calls (hardphones
On Wed, Jan 05, 2005 at 03:56:39PM +0100, Roy Sigurd Karlsbakk wrote:
hi
some time ago, I asked the list of a good book for learning ISDN and
SS7. I don't need to know how to write a channel driver or something; I
just want to know more about the possibilities and what's really sent
back
Guys,
After connecting to the * manager, each and every event is sent to the
connected client, right?
This means that if I install a client on each PC for monitoring incoming
calls, or pretty much anything else, it will create a lot of excess
traffic on my LAN.
Can I connect to the manager and
Title: Message
Yes
yes, we've been through all that actually :-) We did find out it was one of the
3550's reseting the TOS.
-Original Message-From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent:
Tuesday, January 04, 2005 2:40 PMTo:
On Wed, 2005-01-05 at 13:51 +, John Middleton wrote:
Could you please explain or tell me where it is explained the version
and contents of * that is retrieved with CVS.
CVS is sometimes a pain in the but, but it is possible to grab that
information via the log command in CVS.
I am
When I dial *78 I get a 404 error on the phone ( Call failed). Nothing
shows in the Asterisk console.
You need to check dial plan in the 480i...
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
We all mostly know that * as well as various SIP phones support SMS.
While the final setup is somewhat of a mystery, there are reports of
those lucky souls who have it working. We also know that in order to
send an SMS to a mobile phone, we need to connect to some SMS message
center and get
Hi,
Is there some script which can be called from a * extension to
playback the recent incoming
callers on a particular PSTN line?
In the UK 1471 is a BT number which plays back the most recent callers
number, it also
gives you the option to call this number back (now charging you for
this
does TE405P support 3Bit CAS?
what are the configuration tips?
thanx,
Paradise Dove
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hi,
The key to this stuff is using the exact versions of the required libs and
following blindly the instructions (the pwlib and openh323 libraries from
sourceforge.net worked better in my case than the ones from
innaccessnetworks.com). What is the error message you get when you try to
compile
Dear list,
I am starting to setup an asterisk pbx, using a Fritz ISDN card through
chan_capi (0.3.5). The underlying OS is SUSE 9.2; I installed asterisk
with the RPMs supplied on the DVD.
While I can dial out (I had successful outside calls), through the ISDN
card, so far I could not answer a
[EMAIL PROTECTED] is believed to have said:
some time ago, I asked the list of a good book for learning ISDN and
SS7. I don't need to know how to write a channel driver or something; I
just want to know more about the possibilities and what's really sent
back and forth. I was told the book
1) Download latest openh323 Libraries,
untar to a folder and configure them running
./configure
2) Download latest pwlib libraries
3) Download Asterisk-oh323 untar to a folder
Edit the Makefile and configure the folder names that has pwlib
and openh323 libraries
4) make install
The pwlib and
It works with *. But indeed must be installed on a windows client only. This
gateway is user oriented, not server oriented.
Imagine while visiting a customer abroad, you connect to a wifi hostpot with
your laptop (sorry under windows) you take your dect phone or dect earset
only and you can
I have several contacts that use Vonage and was wondering how I can peer
with Vonage (assuming that's possible) so that I can contact these people
through the * rather than PSTN. Can that be done? What about other
providers (Skype, etc)? Is there something on the Wiki that discusses this?
Hi
Yes, Im looking a way to connect Asterisk to HG.
I have already oh323 configured in Asterisk, but I
cant connect to the Siemens HG PBX by ethernet, because the HG doesnt
support normal H.323.
How are you connecting Asterisk with the HG PBX?
Are you connecting thru witch port ?ethernet,
Hello-
I have a client in Orange County California who will soon need some
consulting assistance with their new asterisk system. I've been asked
to help them find someone. Skills needed would be, in order of
importance: Basic experience configuring and using asterisk, coding
experience in
This is a pretty nice looking solution Is anyone else using it? If
so, how is the quality? I do like the idea of keeping the phone lines
where they are and not using the TDM400 series cards.
So far Digium's support has been underwhelming. Today is day 3 since I
tried to contact them. I'm
Hello,
Meanwhile i've downloaded ,again, the 0.6.5 version. I'm using pwlib
and openh323 versions from sourceforge.
It compiled without errors, but the error at startup it's the same
This is the ldd output for the driver.
Shouldn't this be linked to the wrapper ?
# ldd chan_oh323.so
On Wed, 2005-01-05 at 05:59 -0700, Wiley Siler wrote:
Apologies if the format of the email was troublesome. I am accessing my
email remotely via Outlook Web Access otherwise the format would have
been plain text.
Thats good as this message was very easy to read.
Using my analog line for fax
Check out http://www.voip-info.org/wiki-Asterisk+Bootable+CDROM.
/Anders
A while ago, I saw some threads on booting linux w/ asterisk
from a CF card.
I have also seen CD installs of Asterisk, which require a hdd.
Has anyone come up with a bootable cd (like a Live CD), that
creates
On Wed, 5 Jan 2005, Wiley Siler wrote:
So, I need to learn more about voice T1s? Reeally? That would be why I
am posting to the user group in the first place. To learn more. The
wiki says nothing about how PRI works because it is expected that
someone will know. Well, I didn't. Had to
On Wed, 2005-01-05 at 11:00, Mike Dent wrote:
Hi,
Is there some script which can be called from a * extension to
playback the recent incoming
callers on a particular PSTN line?
In the UK 1471 is a BT number which plays back the most recent callers
number, it also
gives you the option to
In article [EMAIL PROTECTED],
Paul Brock [EMAIL PROTECTED] wrote:
Finally, Anyone know of a Digium hardware Reseller in the Uk at all??
www.telappliant.com
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] -
Hello,
Meanwhile i've downloaded ,again, the 0.6.5 version. I'm using pwlib
and openh323 versions from sourceforge.
It compiled without errors, but the error at startup it's the same
This is the ldd output for the driver.
Shouldn't this be linked to the wrapper ?
# ldd chan_oh323.so
Given that all agents are on SIP phones (Cisco 7940), and all outside lines
are EuroISDN channels (using a TE405P), is there any way of finding what zap
channel is being used by an agent channel within a dialplan ?
If you type show agents on the CLI you get information like:
6038 (Agent 6038)
Hi,
You
should connect HG1500 with Ethernet port. But be careful, because in the HG1500
configuration, you need to declare nodes. This node should be the asterisk H323
[EMAIL PROTECTED] The asterisk need to be declare as a gateway, and not as a
phone!
I
never do such config, but as I was
Try http://knopsterisk.com/
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Shahed
Moolji
Sent: Wednesday, January 05, 2005 11:23 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Bootable Asterisk CD ?
Hi All,
A while ago, I saw some
On Monday 03 January 2005 19:34, [EMAIL PROTECTED] wrote:
Has anyone had success using a TE410P card in an HP-Compaq DL380 G4
server?
We're struggeling with the same thing right now. We have several TE410Ps
working on DL380G3s, but have so far been unsuccessful in getting it to work
on the
Anyone with any thoughts I would really appreciate any help.
Having owned one of these for several months, here's what I've learned.
If you don't have a power supply that is stable at 1200ma the IAXy
won't work. It kinda/sorta worked on a 1amp supply but it works more
consistently on a
On Wednesday 05 January 2005 16:25, Raymond McKay wrote:
As of now, the hack still applies. It would be wonderful though if
somebody could implement a command line variable that allows you to append
anything to the SIP URI in the form of variable=variable. Right now the
patch essentially
LOL - Thanks for not getting mad about my email. I just felt a little
stung for being uneducated about T1s but we have to learn somewhere!
I completely understand your concerns and will try to comply as best as
I can.
Again, thanks for being such a contributor to the this support system!!
Jon Radon wrote:
I think you need to subscribe to the context where exten 200 exists.
I'm not sure if it'll work with an arbitrary context. You may also
want to try sending the hint to just one phone. I'm not 100% on the
format for sending the hint to multiple phones.
Just a little bit I got
On Wed, 2005-01-05 at 10:23, Jay Milk wrote:
We all mostly know that * as well as various SIP phones support SMS.
While the final setup is somewhat of a mystery, there are reports of
those lucky souls who have it working. We also know that in order to
send an SMS to a mobile phone, we need to
How can I *accept* messages on my
voip-based US landline?
I doubt it. SMS depends upon the sender and receiver talking via FSK
*before* the phone is answered. I wish fax worked this way, by the
way.
___
Asterisk-Users mailing list
Thx Tony, have dropped them a line :)
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: 05 January 2005 16:41
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Asterisk Pbx Manager Equivalent (in plain
text-
Joao wrote:
Meanwhile i've downloaded ,again, the 0.6.5 version.
I'm using pwlib and openh323 versions from sourceforge.It compiled without
errors, but the error at startup it's the sameThis
is the ldd output for the driver.Shouldn't this be linked to the wrapper
?
I'm having problems
HELP!
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw
register =
[EMAIL PROTECTED]:X:[EMAIL PROTECTED]/1234
[sip.broadvoice.com]
HELP!
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw
register =
[EMAIL PROTECTED]:X:[EMAIL PROTECTED]/1234
[sip.broadvoice.com]
Peter,
I also made it a point to voice my appreciation and recognize the fact
that Stephen is major contributor here. I also acknowledged his
generous explanations. I have also since replied to his reply and
thanked him again as well.
A consultant so I can get a T1 PRI on my wall and use it
What no download? Just wait AsterLinux will be out soon.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jeff R Glassman
Sent: Wednesday, January 05, 2005 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi!
I would like to use lcdproc and asterisk.
Any hints or links? Maybe someone
has experience in such matter. I am working
on such solution. I've heard of SAPBX.
Thanks for any help.
Regards,
Corvin
___
Asterisk-Users mailing list
HELP!
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw
register =
[EMAIL PROTECTED]:X:[EMAIL PROTECTED]/1234
[sip.broadvoice.com]
Rich Adamson wrote:
It is just sending a sip invite to [EMAIL PROTECTED] Does the
X-Lite need to connect to via a proxy?
No. You should work on configuring xlite to register with asterisk.
Thanks. I can get it to work that way. What I was trying to simulate was
an external user calling in.
Dear All ~
I have * setup running ok (with two Wildcard X100P's to PSTN). I also have
two analog phones connected into same through a SIPURA 2000. These work fine,
except that when I call out through PSTN try to send DTMF tones to (say) a
remote PBX to dial an extension, the gain seems to go
Hi Roy,
On Wed, 5 Jan 2005 15:56:39 +0100, Roy Sigurd Karlsbakk
[EMAIL PROTECTED] wrote:
I was told the book ISDN and SS7: Architectures for
Digital Signaling Networks by Uyless Black (ISBN 0132591936) was a
good choice, but this seems sold out. Does anyone know about another
book about the
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