Hello,
how i can tranfer call to another user? Im using X-Lite, i have configured in
features.conf:
[featuremap]
blindxfer = #1
disconnect = *0
automon = *1
atxfer = *2
But when im dial *2 in conversation nothig happens.
What can br problem?
Im using asterisk CVS-HEAD from 02/09/05.
Peder @ NetworkOblivion wrote:
And it's wink-start on an EM analog circuit, not on a standard analog
phone line from your telco. You would need a card that supports EM to
do it even if the telco provided it (not sure if the Digium cards
support it, but I tend to doubt it).
We do not have
Marco Balmer wrote:
Any ideas or hints?
Yes. Whatever documentation told you that you could share a Realtime SIP
peer database between two Asterisk servers was in error (or at least
very incomplete).
There are ways to do it right now, but it's not trivial and does not
provide all the
I have very loud sound through IAX2 channel,very saturated in some
moments.How to find where is problem. I think problem is at provider
side, but how to be doubtless?
Is there any method to measure and change sound level on IAX channel (like
on Zap channel)?
I think, that mistake is between PC and chairs. When i have not outgoing
lines it's too hard to call out. Now i'm in state, that example form
README dialed and i'm trying to receive fax on other side.
Thanks,
Bob.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hello
On Fri, 14 Oct 2005 01:25:20 -0500, Kevin P. Fleming wrote
Marco Balmer wrote:
Any ideas or hints?
Yes. Whatever documentation told you that you could share a Realtime
SIP peer database between two Asterisk servers was in error (or at
least very incomplete).
Server1 acts as a SIP
Multitech makes ATAs and Gateways that support EM signaling:
http://www.voip-info.org/wiki/view/Multitech
- Original Message -
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, October 13, 2005
hi, i facing a problem here. in my sip.conf, i specify a account like this,
[1234]
type=friend
context=from-sip
username=1234
secret=1234
nat=no
canreinvite=yes
dtmfmode=info
[EMAIL PROTECTED]
disallow=all
allow=ulaw
so i am able to login with username 1234 and password 1234
but ther weird part
On Thu, 2005-10-13 at 17:02 -0700, Bruce Ferrell wrote:
Hi all,
Trying to build ztdummy on an old redhat 7.3 box running kernel
2.4.20-43.7.legacysmp. Yes, I have the kernel sources installed. Yes,
I set them up with make oldconfig; make dep.
The build error is:
make ztdummy
gcc
Hello everybody,
I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip
installation.
I actually have a problem when a PSTN user calls and hangs up. The
disconnect tone is not
detected by the SPA, the the call continues and, for example, leaves an
empty message on
the
Title: Patrick Briefpapier
All,
Currently I've got
my Asterisk machine running smoothly on IP bases. Meaning I can reach all phones
or softphones within my LAN or remote LAN's via VPN. The next step for me is
connecting it to the PSTN network.
After some tweaking
with the modem.conf I got
On 10/14/05 15:42 Kong said the following:
but ther weird part is, i can also register as any number (account)
without having to specify in sip.conf. thus anybody can just use my
under the [general] section, use a context which limits what
unauthenticated users can do/call. it can even be
On Fri, 2005-10-14 at 10:15 +0200, Patrick de Kok wrote:
All,
Currently I've got my Asterisk machine running smoothly on IP bases.
Meaning I can reach all phones or softphones within my LAN or remote
LAN's via VPN. The next step for me is connecting it to the PSTN
network.
After some
how to chech if the user is an unauthenticated one? thank you
At 03:58 PM 10/14/2005, you wrote:
On 10/14/05 15:42 Kong said the following:
but ther weird part is, i can also register as any number (account)
without having to specify in sip.conf. thus anybody can just use my
under the
I have a telephone Voismart IP PHONE 106.
I have lost the password of the telephone and
therefore I am not able to set up it. How can I do to
do a reset of the telephone?
___
Yahoo! Messenger: chiamate gratuite in tutto il mondo
Are there any configuration options to allow certain sip/iax accounts
to dial out over specific trunks, and also to stop them dialing out over
other trunks.
Thanks in advance
Bails
___
--Bandwidth and Colocation sponsored by Easynews.com --
On Thursday 13 October 2005 15:20, Apu Islam wrote:
Is DID on analog line possible ? ( my telco is qwest) . Just wondering if
there is any way to test it on anlog wcfxo cards.
Another approach is to use a CTPX or Exacom unit to convert the DID or
2-Wire EM signal into a signal
On Mon, 2005-10-03 at 17:54 -0400, Matt Roth wrote:
List members,
2) What will happen on the NFS client if the NFS server crashes (I expect the
leg files to be written to the local mount point until the mount is
reesablished)?
Why don't you create a file on the NFS server called something
On 10/14/05 16:40 Kong said the following:
how to chech if the user is an unauthenticated one? thank you
read www.voip-info.org on SIP.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
I want to add H323 support to my asterisk setup. What are the pros and cons of
the available modules, h323, oh323 and ooh323 and which is the best one to go
for?
Obelix
This message was sent using IMP, the Internet Messaging
Title: Patrick Briefpapier
Some additional
information:
mchan_modem.so[0;37;40m] =
([33;40mGeneric Voice Modem Driver[0;37;40m)Parsing
'/etc/asterisk/modem.conf': FoundLoading modem driver chan_modem_i4l.so
= ([33;40mISDN4Linux Emulated Modem Driver[0;37;40m)Configured modem
can i know where to start? SIP is such a big topic.
At 05:58 PM 10/14/2005, you wrote:
On 10/14/05 16:40 Kong said the following:
how to chech if the user is an unauthenticated one? thank you
read www.voip-info.org on SIP.
--
Regards, /\_/\ All dogs go to
hi
can i make sip three way call on asterisk
i meen one person call one time to two another
and when they answer this 3 person speak with each other
as in confereces
i cant use
meetme becouse i need send dtmf
--
Oleh Mukha
IClub
380322722738
www.ic.lviv.ua
Hi all,
I have 12 SIP phones at a particular site all connected to a local asterisk
server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming
calls. An IAX gateway is used for outbound calls. At the moment, when an
incoming call comes in, asterisk dials every SIP phone like so:
Hello,
I have set up 2 different fwd.pulver.com accounts on my Asterisk. One will ring all my phones through one context, while the other account was set up to fool Nigerian scam artists, and will go directly to a special voicemail (after a few rings to give the impression of ringing a real
Hello,I am trying toconfig inter Asterisk IAX2 connection.
When I register a username and password it works but I would like that "Any" incomming SIP call (without specific username and password) pass throught IAX2 for delivery to the other end *.Is it possible ?I read in Asterisk IAX config, if
makevuy wrote:
Hello everybody,
I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip
installation.
I actually have a problem when a PSTN user calls and hangs up. The
disconnect tone is not
detected by the SPA, the the call continues and, for example, leaves an
empty
All works very well. Last question is if there is a chance to get result
of sending by mail (for example as answer to my mail).
Thanks,
Bob.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
aka Bret McDanel
Sent: Thursday, October 13, 2005 12:03
[EMAIL PROTECTED] wrote on 10/12/2005
01:23:57 PM:
On Wed, Oct 12, 2005 at 12:05:32PM -0400, [EMAIL PROTECTED] wrote:
I am in the middle of trying to get a milliwatt test line to
calibrate the
rxgain properly. However, this won't help me with the txgain,
will it?
How can I properly
Thanks. I'm prevented from testing it right now, but I will as soon as
possible. It seems to be the fix that I need.
Lars.
On 10/13/05, Matt [EMAIL PROTECTED] wrote:
Try disabling inband call progress tones. Let Asterisk handle everything.
In sip.conf add the line:
progressinband=no
On
Hi!
I have installed two hfc-s cards to handle my pstn calls. I use mISDN with capi,
so capi.conf is edited. I have tested both separate and cards are working well.
But they are not working together. It seems that when i set up settings for the
other card:
;capi.conf:
[general]
The phone carries their configuration from the TFTP server, regarding the
manufacturer.
You should be able to change the password from the configuration file on
the TFTP server.
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fabio
This is probably a really bad question to ask but here goes. Does
asterisk work with any of the T1/E1 cards from SBE? I'm sure SBE is
a competitor to Digium, but I have access to SBE cards and the linux
driver. Just curious more than anything. Thanks.
--
Michael J. Lynch
What if the hokey
Where can I find this information?
Faris Raouf wrote:
makevuy wrote:
Hello everybody,
I'm a new user of * and I just bought a Sipura SPA-3000 to make a
home voip
installation.
I actually have a problem when a PSTN user calls and hangs up. The
disconnect tone is not
detected by the SPA,
Bails wrote: -
Are there any configuration options to allow certain sip/iax accounts
to dial out over specific trunks, and also to stop them dialing out over
other trunks.
Thanks in advance
Bails
=
Bails,
Set the extensions to use certain
Title: Patrick Briefpapier
I would prefer to
get it working with i4l at the moment, and migrating later on to CAPI if
needed.
Thanks for any help
you can give me..
-
Patrick
---
This email was scanned by MyMail of DatacomPartner
George Pajari wrote:
On Thursday 13 October 2005 15:20, Apu Islam wrote:
Is DID on analog line possible ? ( my telco is qwest) . Just
wondering if
there is any way to test it on anlog wcfxo cards.
Another approach is to use a CTPX or Exacom unit to convert the DID or
2-Wire EM signal
Marco Balmer wrote:
Server1 acts as a SIP Client only. Server2 should act as a SIP-Server with the
sip_buddies table on the MySQL-Server.
But this is not currently implemented. There is a patch in the bug
tracker that will help move in this direction, but it's only a start,
there are many
I asked the same to Ben Dewey (SBE) a couple of weeks ago, and I get no
answer.
As I have a couple of cards, and I know that I can do channelized with those
card, I believe that all that I should do is try it.
If you know something different, let us know.
Thanks,
Carlos Alperin
-Original
What do you mean with 'not working'?
Do you get any error messages? What does the log show?
Do both cards work without asterisk/chan_capi?
Armin
On Fri, 14 Oct 2005 [EMAIL PROTECTED] wrote:
Hi!
I have installed two hfc-s cards to handle my pstn calls. I use mISDN with
capi,
so capi.conf
Michael J. Lynch wrote:
This is probably a really bad question to ask but here goes. Does
asterisk work with any of the T1/E1 cards from SBE? I'm sure SBE is
a competitor to Digium, but I have access to SBE cards and the linux
driver. Just curious more than anything. Thanks.
SBE does not
Has anyone figured out what this message means:
Don't know what to do if second ROSE component is of type 0x6
We are running a PRI through a Sangoma card that is handling the
D-channel natively at this point, but we go the error when zaptel was
handling the D-channel, too. I have googled, but
Hi,
I don't if was yet an issue.
It really would be nice if each user is able to active/deactivate the mail
forwarding of his voicemail via the VoiceMailMenu.
Regard
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
I have been receiving a lot these 488 Not Acceptable Here from a number of
providers. What could the problem be?
What is the most common cause of this message?
This message was sent using IMP, the Internet Messaging Program.
My Asterisk PBX seems unable to receive DTMF information via SIP. I have tried
all the various methods, rfc2833, inband and info and they all don't seem to
work. IAX2 works fine. Is there something I must be missing
?
/Obelix
Kong wrote:
can i know where to start? SIP is such a big topic.
Try looking for SIP configuration (sip.conf) in the Wiki, it's got lots of
examples. Or you can also try looking it up on google.
Flynn
___
--Bandwidth and Colocation sponsored by
On Fri, 2005-10-14 at 15:17 +0200, Patrick de Kok wrote:
I would prefer to get it working with i4l at the moment, and migrating
later on to CAPI if needed.
Thanks for any help you can give me..
And the large number of answers you have received on how to make i4l
work doesn't say
Hi zoltan,
I have got the same problem...same error. Seems like the makefile is
searching for a modules rule but I looked into Makefile and there is
not a 'modules' rule...
Have you found a solution?
TIA
Giorgio
Zoltan Szecsei wrote:
Bob Goddard wrote:
On Friday 01 Jul 2005 15:14,
I'm running into an issue where subscribers to our service cannot call certain
1-800 numbers if they have a caller id blocked account (restrictcid=yes).
This is on Asterisk 1.0.9 and our clients are using Sipura SPA-2002's.
Our provider uses a SIP/PSTN gateway, so we hand off SIP to them from
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Nickerson
Sent: Wednesday, October 12, 2005 9:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Which asterisk-friendly cards
are fax-capable?
Perhaps they dont' like the codec you're offering in your INVITE message?
Ray
On Fri, Oct 14, 2005 at 01:36:17PM +, Obelix wrote:
I have been receiving a lot these 488 Not Acceptable Here from a number of
providers. What could the problem be?
What is the most common cause of this
Hi,
I've set up IAX FreeWorldDialup on my asterisk server but when I dial my
number, I get message '486 Busy Here '. When I dial any other number from my
*, it says 'All Circuits are busy now'. What is the problem with my
settings? I've followed all the instructions step by step.
Hector
Hi,
When I try to load chan_iax2.so, I get the error message
The channel name is iax. Yet it provides commands such that begin with
iax2 and listens on port 4569.
??? In /usr/lib/asterisk/modules the name of the file ist chan_iax2.so
and as far as I understood, I have to enter
Hi,
I'd like to add a feature to my asterisk system
that tries to find a user among a couple of locations, and then goes to internal
voicemail if the user doesn't pick up. (e,g, an internal extension and a
cell phone). The catch is that I want the user to manually accept the call
to
I get audio from mpg123 at the command line but when I load up asterisk
and try to get audio from the console it looks like it's working, and
even pauses like it is playing the file but there is no audio coming
from the speakers.
I have searched and looked through the archives and tried to fix
FWIW, we are also seeing this message each time we receive a call. I
also went the Google route and found only questions, not answers. We
are running a PRI from US LEC (channels 1-10 are B-channels, with
channel 24 being the D-channel, and we are only running voice on the
PRI.) The PRI is
Anyone know where to get a reasonably priced/chat PoE (powered) switch?
For about 5-12 ports?
-- Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Thursday, October 13, 2005 8:56 PM
To: Asterisk Users Mailing List -
I havecoded a new application in Asterisk called app_followme that will do what you're looking for. The caller who made the call originally is also optionally put on hold music while the hunt is going on. There's also planned functionality for blacklisting certain callerIDs so a caller who is
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jeremy Gault
Sent: Friday, October 14, 2005 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Don't know what to do if second
ROSE component is of type
New Language syntax for CVS-HEAD
Set(LANGUAGE()=language)
How do I use that exactly?
Set(LANGUAGE()=en) is for English
Set(LANGUAGE()=fr) is for French
How do I set it up for Chinese?
Set(LANGUAGE()=xx) where do I get the xx ???
The *.gsm files goes for each language into:
I have been getting that message also. I have been using various
versions of CVS head since Feb. 2005.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
Gault
Sent: Friday, October 14, 2005 10:23 AM
To: Asterisk Users Mailing List -
Quoting Ray Van Dolson [EMAIL PROTECTED]:
How can you determine which codecs are acceptable to them?
Do they have a way of indicating it?
Perhaps they dont' like the codec you're offering in your INVITE message?
Ray
On Fri, Oct 14, 2005 at 01:36:17PM +, Obelix wrote:
I have been
I also have the same problem at a customer. It's PRI setup as 5ESS.
It doesn;t seem to be hurting anything but it is related to the caller
ID. I tried google for docs etc but I came up with nothing I could
understand or figure out.
Over time I've used both Sangoma and Digium card and it
Dave Cotton wrote:
snip
The above is pointing to USB devices are they configured in the kernel
and does the machine support uhci?
Suggestions?
2.6 ztdummy does not need USB devices.
gcc 2.96 don't know if that also will have probs long time since I used
it.
Upgrade it's free.
Yes. They tell you what's acceptable to them inside the SIP messages you're trading back and forth on the call setup.
You can use sip debug within Asterisk to get a closer look at those messages.
On 10/14/05, Obelix [EMAIL PROTECTED] wrote:
Quoting Ray Van Dolson [EMAIL PROTECTED]:How can you
Apparently ENUM now REQUIRES a + at the beginning of the number to query.
EnumLookup(+18886532145)
No I didn/t see it documented anywhere.. It seems to require it even if
there is no + at the beginning of the ENUM record in DNS. We use ENUM
to look up extensions and so should not prefix the
Obelix wrote:
I have been receiving a lot these 488 Not Acceptable Here from a number of
providers. What could the problem be?
What is the most common cause of this message?
In my experience, that is caused by one side requireing a codec that the
other side does not support.
BJ, thanks alot for the coding but I see no reason for it as all you
mention is doable currently in Asterisk using some DP magic.
Will just search the list and you should find some examples on how to
do it (I think there is even an example on the wiki try:
SIP DEBUG would show the information in the INVITE.
Obelix wrote:
Quoting Ray Van Dolson [EMAIL PROTECTED]:
How can you determine which codecs are acceptable to them?
Do they have a way of indicating it?
Perhaps they dont' like the codec you're offering in your INVITE message?
On Fri, 2005-10-14 at 08:24 -0700, Bruce Ferrell wrote:
She's an old dual proc MB and I've had much difficulty doing upgrades
else I would.
maybe it's time to do the MB upgrade but there is much pain and expense
in doing so.
There comes a time... My HP NetServer 5/100 LC looks like
I need to pick all the Asterisk and Cisco People a little.
My company has a Cisco Call Manager 3.3, configured via h323 gateways. I
have remote users that I want to place a SIP Server on the external WAN
and be able to connect their phones to the system and be able to get calls
and call people in
In Peru you can request Telefonica to provide reversal polarity.
Jorge
makevuy wrote:
Where can I find this information?
Faris Raouf wrote:
makevuy wrote:
Hello everybody,
I'm a new user of * and I just bought a Sipura SPA-3000 to make a
home voip
installation.
I actually have a
Dave Cotton wrote:
On Fri, 2005-10-14 at 08:24 -0700, Bruce Ferrell wrote:
She's an old dual proc MB and I've had much difficulty doing upgrades
else I would.
maybe it's time to do the MB upgrade but there is much pain and expense
in doing so.
There comes a time... My HP NetServer 5/100
You can use H.323 on Asterisk and setup CM to use an H.323 gateway to
Asterisk, or setup a gatekeeper and have both ends talk to the
gatekeeper. I have redundant CM boxes, so I HAD to use a gatekeeper and
set it in proxy mode because I had media path issues (the call initiated
from one box,
Kib Eki wrote:
It really would be nice if each user is able to active/deactivate the
mail forwarding of his voicemail via the VoiceMailMenu.
Also, one simple thing. Is it possible to listen to my greetings
without having to re-record them?
___
I am in the middle of trying to get a milliwatt test line to calibrate the
rxgain properly. However, this won't help me with the txgain, will it?
How can I properly calibrate the txgain? By ear? Or is there a more
scientific method?
I contacted Rhino to see if they had any
Tom Rymes ha scritto:
Has anyone figured out what this message means:
Don't know what to do if second ROSE component is of type 0x6
IIRC, it has to do with rerouted/forwarded calls. I came across that
portion of source code when I dealt with call forwarding/deflection.
ROSE stands for
Salve Fabio,
volevo dirle che può contattare il supporto voismart
a [EMAIL PROTECTED]
Nel caso non avesse ricevuto risposta, mi avverta
che provvederò io stesso a farle avere informazioni.
(lo farei ora, ma sono sul treno e non ho accesso ai miei dati)
:)
greetings,
Matteo brancaleoni.
On Fri,
Does anyone know what this error means?
WARNING[12632]: chan_zap.c:10084 setup_zap: Ignoring signalling
Thanks,
Andy
--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]
Does anyone know why you must sometimes dial a number twice to get it to
connect? We have IP phones and a TDM400P Digium card that connect to 2 PSTN
lines. Some numbers work fine and then other don't.
What gives?
Thanks.
___
--Bandwidth and
Does anyone Know if tha avaya 4621 IP phone work wiht asteisk?
if it work it has featuras working
Thanks
Ignacio
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
BJ,
Great news - glad to hear it.
I think the key thing I'm looking for is that this is all transparent to the
caller. I want them to hear nice hold music while the user is searched for,
and only be directed to the physical extension if the correct person picks
up. (e.g. no third party
The file used with a Digium IAXy device: iax.conf
has the line: port=5036 (I also use the bindaddr=192.168.1.91 entry)
but when Asterisk talks to the IAXy device it
used port 4569 (from tcpdump). How are the port numbers assigned?
What tells the IAXy which port to use. The IAXy provisioning file
On Fri, Oct 14, 2005 at 11:38:00AM -0600, Rich Adamson wrote:
A milliwatt generator creates an audio signal at 1,004 hz and 0db. It
has nothing at all to do with a T1 signalling, etc. You can yell into
a analog telephone set and create audio levels greater then 0 db.
Whoever is feeding you
Does anyone know why you must sometimes dial a number twice to get it to
connect? We have IP phones and a TDM400P Digium card that connect to 2 PSTN
lines. Some numbers work fine and then other don't.
What gives?
The most frequent cause that I've heard of is that asterisk starts pumping
Paul wrote:
Last time I checked analog DID trunks were expensive both NRC and MRC.
Depends on your ILEC/CLEC. Here is Vancouver they are the same price as
non-DID trunks with DID numbers $2/ea in quantities 1000 (from at
least one CLEC). I have heard of CLECs in the US where DIDs are a
Michael Van Donselaar wrote:
On Thu, 13 Oct 2005 08:41:17 -0400, Paul [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
Hi,
Can anyone recommend a USB phone that can be used under Linux, either
interfacing directly with Asterisk in some way, or using a soft phone
program on Linux
Hello all,
is here anybody who have any idea how i can insert a script or program to
rewrite a callerid with special rules. This ist necessary because of many
moving mobile offices who changes the telefonenumbers in short time
distances. I've found the SetCIDName feature. But this doesn't work
Am Freitag 14 Oktober 2005 21:22, Hans-Peter Straub schrieb:
Hello all,
is here anybody who have any idea how i can insert a script or program to
rewrite a callerid with special rules. This ist necessary because of many
moving mobile offices who changes the telefonenumbers in short time
Hello All,
I am trying to use soxmix to merge two wav files
generated by monitoring calls from a queue, since it generated two files (in
out).
When I run soxmix file1.wav file2.wav
mixedfile.wav, although file1.wav and file2.wav are good, mixedfile.wav is file
with the same size as
I recently upgraded
my Asterisk system to the latest CVS-HEAD
Asterisk CVS HEAD
built by[EMAIL PROTECTED] on a i686
running Linux on 2005-10-12 13:34:09 UTC
Ever since this
upgrade, the system is jumping n+101 if it gets a busy on a Dial command, it is
now proceeding to the next priority
On 10/14/2005, William M. Sandiford [EMAIL PROTECTED]
wrote:
I recently upgraded my Asterisk system to the latest CVS-HEAD
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-10-12 13:34:09 UTC
Ever since this upgrade, the system is jumping n+101 if it gets a busy
Hi,
Try add
[1234]
...
host=dynamic or host=xxx.xxx.xxx.xxx (the client's IP)
...
...
AK
On 10/14/05, Kong [EMAIL PROTECTED] wrote:
hi, i facing a problem here. in my sip.conf, i specify a account like this,[1234]type=friendcontext=from-sip
I got my next cabinet convinced to use his digital PRI to do the test, which would be what we would really use after all. I have an idle digum T1 card sitting, so lets see if I can get that going.
Thanks for all your help. I learn something new everyday.
-apu
On 10/14/05, George Pajari [EMAIL
Hi list!
I´m connecting a Brasilian voip (- gvt.com.br -) provider through my
asterisk box and using the register command from sip.conf. What I can´t
understand is why my unit sends a new registration message every minute!
And every time my asterisk box sends a registration, it gots a
[EMAIL PROTECTED] wrote:
On 10/14/2005, William M. Sandiford [EMAIL PROTECTED]
wrote:
Ever since this upgrade, the system is jumping n+101 if it gets a busy
on a Dial command, it is now proceeding to the next priority (n+1)
Has something changed with this? Is there a way to change it
George Pajari wrote:
Depends on your ILEC/CLEC. Here is Vancouver they are the same price
as non-DID trunks with DID numbers $2/ea in quantities 1000 (from at
least one CLEC). I have heard of CLECs in the US where DIDs are a
tenth of this cost.
Yep. We're using US LEC here (with a PRI)
Hi all,
Im trying to build an small home system. I have 2 pots lines, and i need to make 8 extensions and be able to use my old analog phones.
What would you recommend to use asthe 8FXS switch?
I saw some equipment from quintum, they have a Tenor AS that offer 4 FXS ports. But i don't know if it
Ever since this upgrade, the system is jumping n+101 if it gets a busy
on a Dial command, it is now proceeding to the next priority (n+1)
Has something changed with this? Is there a way to change it back?
So glad to see you read the documentation...
Try scanning UPGRADE.txt
I dont think the Quintum hardware
supports SIP devices (just SIP trunks).
-Jonathan
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Claudio Canseco
Sent: Friday, October 14, 2005
4:32 PM
To: Asterisk Users Mailing List -
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