In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Does this belong to my dialplan or my sip registration settings?
To your SIP registration settings. You should limit that user/peer/friend to
only one line.
--
Tomislav Parcina
tparcina#lama.hr
[EMAIL PROTECTED] wrote:
Looks very nice.. Is it GPL, GNU?
PBXware interface is not GPL/GNU currently.
Some time in the future we may release is it under GPL/GNU license :)...
___
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Asterisk-Users
Anthony,
I will suggest you to use E1, you got 30 channels to communicate. I did the
integration with Toshiba CTX using E1, and no problem at all.
Asterisk as Pri_net
Toshiba as PRI_cpe
/etc/zaptel.conf
span=1,1,0,ccs,hdb3 ;this is depanding on your pbx settingyou got to
test it out,
I have an AGI program with an array containing a set of ${UNIQUEID}
variables for channels that may be active on the system. I need a way
for the program to tell if they are or not.
It's certainly possible using the manager interface, or appropriate
asterisk -rx commands, but I'd prefer to do
On Thu, Mar 02, 2006 at 04:14:34PM -0700, Douglas Garstang wrote:
The best way to achieve maximum manageability is to design a MySQL database
and develop AGI scripts (in your language of choice) that work to that
design. I've found that it has been far easier to develop complex routing
So, simply respawning asterisk, or checking to see if it's running
isn't good enough, because asterisk is indeed running. We need to
access asterisk and issue a command, and see if asterisk responds
appropriately. If not, we can assume it has died, and we can kill it
off (killall -9
Hi all
I'm a newbie in asterisk.I install asterisk server
successfully. I configure this server to traverse NAT.
Using Xlite clients, i make a call between 2 local
networks through Internet.Asterisk server is
installed on a host with public IP. client A (in the
LAn A) and client B (in the LAN
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
what are the file permissions/ownership and are they readable by the
asterisk process ?
The problem was that wav files where in stereo mode. I have encode them and now
it works fine.
--
Tomislav Parcina
[EMAIL PROTECTED]
I know there's a variable for the IP of a SIP channel, but I can't find
if such a variable is avaliable for a generic voip cahnnel, or at least
h323 channels (ooh323)
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Hi,
I'm looking for someone to do time-to-time mantainence on some of our
machines going up in New York. The person *MUST* be stationed in New
York.
Areas of expertise required:
- Proficiency in Linux: Slackware, Fedora
- Proficiency on Cisco Routers
If anybody is interested, please
ram wrote:
Hi
how about when trying to call SIP extention to SIP extension
Local cal
even though its going to out route
when i enable SIP_IAX=YES
then its IVR in place ask 9 to dial SIP/IAX, if not its dial to
international call
How can i avoide this
check if the user belong to
hi all i have a asterisk configured and working perfectly. but i have a
problem.
if i download a softphone for example sjphone and digit for example
[EMAIL PROTECTED] i receive this call. is possible to block this?
i only want to received calls for login users...
--
.-
hi
if i have an agents that figure as a member in more than one queue,
how can i login / logout him in a specific queue an not in all queues?
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To UNSUBSCRIBE or
Hi,
I just want to implement Music on Hold while my * tried to get the
connection on the trunks. Any clues would be welcome...
It needed for me as when failover trunk sometimes take effect, it
takes sometime for the connection and unknowingly we disconnect the
call. MOH is just to avoid this.
On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Mar 2, 2006, at 9:46 AM, Matt wrote:
Doesn't it seem absurd to go through all these gyrations, rather then
troubleshooting and fixing the problem? I know you have already tried
without success, but this seems absurd to me.
I am
On 3/3/06, Tomislav Parčina [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Does this belong to my dialplan or my sip registration settings?
To your SIP registration settings. You should limit that user/peer/friend to
only one line.
How is the best way to
This has to be the worst documentation I have ever come
acrossed. I have found two or three docs on how to install it, but they are all
so different and make huge assumption about what packages you have installed
and locations of files. Has anyone seen something better, I want to get this
First things first... use the latest version... (that I know of)
http://www.fitawi.com/Asterisk/
second... which part are you having problems with? The web piece? or
the app_cbmysql?
For the app_cbmysql, I have found that the easiest way to work with it
is to incorperate it into
Hi All,
I want to configure fax with Asterisk and I found that we can do this reliably
using G711 codec only. Currently my provider is supporting G729 and G711.
During the call initiation the call starts with G729 (1'st priority) and
somehow if the receiver is unable to receive call then we are
I haven't tried sip yet... been finishing voicemail, but the principal
is the same.
res_mysql.conf
[general]
dbhost = localhost
dbname = asterisk
dbuser = someuser
dbpass = somepass
dbport = 3306
dbsock = /var/run/mysqld/mysqld.sock
extconfig.conf
voicemail = mysql,asterisk,voicemail
; i would
Hi,
Is there a command (to use in a dial plan), to check the
call status during a call.
Kind Regards,
Arjan Kroon
Mobillion B.V.
Copernicuslaan 30
Postbus 554 / PO Box
554
6710 BN Ede
tel: +31 (0)318-648920
fax: +31 (0)318-648839
mobile: +31 (0)6-55871460
email: [EMAIL
[EMAIL PROTECTED] wrote:
Hi All,
I want to configure fax with Asterisk and I found that we can do this reliably
using G711 codec only. Currently my provider is supporting G729 and G711.
During the call initiation the call starts with G729 (1'st priority) and
Faxing via VoIP is not reliable
Sean Cook wrote:
First things first... use the latest version... (that I know of)
http://www.fitawi.com/Asterisk/
second... which part are you having problems with? The web piece? or
the app_cbmysql?
For the app_cbmysql, I have found that the easiest way to work with it
is to incorperate
When you type 'help' from the CLI, it says nothing about 'quit' - or at
least not between 'no debug channel' and 'realtime load'. Google told
me about it, and I probably should have guessed, but still...
Who do I report this to?
Bob McDowell
___
Hello,
For a whole lot of different reasons, I am thinking of moving from pure VoIP
(my DID provider gives me SIP access and my termination is SIP too) to PRI
(possibly keeping termination in VoIP for long distance). FYI, my business
is Hosted PBX...and my end-points will stay SIP.
Here is the
Sorry. Miss type 'can'. I meant 'cannot'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Martin
Joseph
Sent: Friday, March 03, 2006 12:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: G729 and Meetme
On
Hello all!
On Thu, Mar 02, 2006 at 03:12:05PM -0600, Joseph Tanner wrote:
The problem isn't that asterisk isn't running, it's that asterisk is
not responding. When asterisk is in this funky state, I can still run
asterisk -r from the command line and get access to the CLI.
While in the
I have not (yet) had one of my bare bones systems lockup.. but they
also don't do 400-700 calls a day.
The system that does lockup experiences exactly the issues described
here which are you can connect to it asterisk -r and issue commands
but nothing responds not even stop now... and you have to
The instructions on the wiki for asterisk Realtime give the extensions
schema with the priority field set to be tinyint(4). This of course
cannot hold the value 'hint'
The question I have, is the solution simply to set the field to
varchar(n) as that will then hold 'hint' or any integer
Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server?Amaury Rodríguez http://liberadospucmm.blogspot.com http://groups.msn.com/telematicaPUCMM2002 Go OpenSource And Be FREE!!
Yahoo! Mail
Bring photos to
Hello there,
I'm successfully using Asterisk Realtime to access information about
voicemail users from a MySQL database. Now I'd like to read static
voicemail information (such as format, serveremail, etc.) also from a
database. Is that possible? If so, I'm assuming one would need to
Does anyone have a good resource to learn how to program the
soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmwarethanks.
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To
A lot of phones support LDAP for this purpose. I don't think X-Lite does, but I've never checked.On 3/3/06, Amaury Rodriguez
[EMAIL PROTECTED] wrote:Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server?
Amaury
Hallo!
I have problem with incoming calls on 2 phone
numbers registered on same SIP provider account. I've tried averything and
nothing seems to work. No matter what I do asterisk system refuses differ betwen
them and both got connected to the same extensions.
I've tride with:
A lot of phones support LDAP for this purpose. I don't think X-Lite does, but I've never checked.On 3/3/06, Amaury Rodriguez
[EMAIL PROTECTED] wrote:Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server?
Amaury
I will
try to test your adaptation.
How I
congfigureto enable VAD?
Regards
Jsalas
-Mensaje original-De: Moises Silva
[mailto:[EMAIL PROTECTED]Enviado el: Friday, February 17,
2006 11:26 AMPara: Asterisk Users Mailing List - Non-Commercial
DiscussionAsunto: Re:
Cisco has a book about it: http://www.amazon.com/gp/product/1587050609/sr=8-1/qid=1141401215/ref=pd_bbs_1/103-7257053-6939020?%5Fencoding=UTF8
While this isn't specifically about the SIP image, the XML browser is the same. I also use Cisco::IPPhone
I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811..
On 3/2/06, Johnathan Corgan [EMAIL PROTECTED] wrote:
Gary Richardson wrote: Now it seems that if I'm really
Anyone have any luck RMAing a Sipura phone since the Cisco take over?
Sipura only has support via email or fax to end users and I haven't
gotten a response to either for over 2 months.
Linksys Support will jump you through all their scripted hoops to
resolve your problem (they hope if they speak
i am taking overr the administration of an existing production * PBX but i cant seem to find out which version of * this is. When i use the 'show version' coomandat the cli, i get this:Asterisk CVS-HEAD built by [EMAIL PROTECTED]
on a i686 running Linux on 2005-08-10 05:57:59 UTCcan anyone tell
I've got a problem with chan-misdn
I'm using asterisk with a hfcsusb-adapter in nt-mode connected to an
isdn-telephone
making calls to other internal clients like sip or sccp are without
problems
if I call into (or receive a call from) the pstn via a zap-channel (Digium
E1-card) my outgoing
On a 55 station
install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are
complaiining about echo. According to the users, the echo seems to be phone
number dependant. They claim that certain phone numbers have echo while others
dont. Are there any tuning parametes like
Is anyone using asterisk with the Zapata hardware and a Sprint FNTM(I
have seen this spelled fantom) T1 line?
I can take calls, but:
1. I don't know the correct way to get ANI/DNIS (*ANI*DNIS*)
2. I cannot place outbound calls.
From a custom app on another hardware/software platform I know from
Dumpolid Exeplish wrote:
i am taking overr the administration of an existing production * PBX
but i cant seem to find out which version of * this is. When i use the
'show version' coomandat the cli, i get this:
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
Can someone please tell me if it's possible to select the G726 codec
bandwidth for an IAX trunk between two Asterisk 1.2.5 servers?
I can select disallow=all / allow=g726 but I think it defaults to the
g726-32 variant.
Is there any way of forcing Asterisk to use g726-24 for such a trunk
On Thu, Mar 02, 2006 at 04:14:34PM -0700, Douglas Garstang wrote:
The best way to achieve maximum manageability is to design a MySQL database and
develop AGI scripts (in your language of choice) that work to that design. I've
found that it has been far easier to develop complex routing logic
In Asterisk the Agent / Queue setup is kinda different than most people may
expect. You can use a Queue without using Agents and Agents can be used without
Queues. Agents however extend normal channels with the ability to
login/logout/pause that is not available on Zap/SIP/IAX/etc.
I assume
I had the same issue with issue with Sipura. I went though email
support. They finaly said email another address to get a RMA. That
support made me go though everything the 1st one did again. And in the
end they never responded after they decided I needed a RMA.
So it never got resolved.
Whisker, Peter wrote:
Can someone please tell me if it's possible to select the G726 codec
bandwidth for an IAX trunk between two Asterisk 1.2.5 servers?
I can select disallow=all / allow=g726 but I think it defaults to the
g726-32 variant.
Is there any way of forcing Asterisk to use
only for the whole cardthe tx and rx gain affect all 24 channels.
-D
From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Fri 3/3/2006 11:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Echo Cancelation
Its good for us to post thing about different companies customer
service. I feel one of the most important points when buying a product
is if the company is going to stand behind it. I had purchased some
sipuras that did not work but was lucky and able to send them back to
the store I ordered
On Fri, 2006-03-03 at 10:45 +0100, serge messa wrote:
Hi all
I'm a newbie in asterisk.I install asterisk server
successfully. I configure this server to traverse NAT.
Using Xlite clients, i make a call between 2 local
networks through Internet.Asterisk server is
installed on a host
Aaron Daniel wrote:
Thanks :) When we were using Mark2 with aggressive suppression, we had
zero problems, but decided to go with the hardware canceler in our new
gateway since hardware's supposed to be better than software...
hopefully this works for us too.
'aggressive suppression' is
Matt Schulte wrote:
All, I'm not sure how to word this question but we're noticing a lot of
our asterisk boxes no longer have multiple asterisk child processes.
i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used
to seeing 8+ .. There is no rhyme or reason to it, and we're
Jordan-
I'm not sure if you found the files and instructions
on www.fitawi.com/Asterisk/. If you did I can offer you a
full refund of the purchase price. (oh right, it's free, I forgot)
I'm afraid I did make some assumptions about which packages
are installed on the system. If
On Fri, Mar 03, 2006 at 11:39:49AM -0600, Kevin P. Fleming wrote:
Matt Schulte wrote:
All, I'm not sure how to word this question but we're noticing a lot of
our asterisk boxes no longer have multiple asterisk child processes.
i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally
I'm doing an install for a client with the following requirements.
- 1 Million minutes of outbound calling
- Calls come in to asterisk via SIP/IAX and terminated to third party
provider via SIP
- Codec usage will be about 70% g711 30% g729 (there should be no transcoding)
- 100% IP setup with no
Does anyone know if there is a way to continue in the dial plan for the
called (outbound) channel if the caller channel disconnects? Something
like this:
*
[call_client]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1},30,g)
exten =
Guys.
I have a te100p with unicall and an E1 and Im having problem with DTMF tones
but the weird thing is, I only have problems sending the tones to certain
phone numbers, anybody seen this behavior?
Asterisk shows on the console the dtmf tone been pressed but seems the other
side is not getting
Hi Ron!
Well, I dont have raid involved here but do use SATA. I
have 14 sip phones and an average of 5 calls at a time. I was thinking about
recording in gsm to a ramdrive and then copying the files to the disk at certain
intervals.
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Dumpolid Exeplish wrote:
i am taking overr the administration of an existing production * PBX but i
cant seem to find out which version of * this is. When i use the 'show
version' coomandat the cli, i get this:
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-08-10
Michiel van Baak wrote:
On 13:02, Wed 01 Mar 06, Arne Morten Johansen wrote:
Hi there.
Is it possible to have different sip users have the same CallerId number
in sip.conf.
I need this because we got multiple companies on this Asterisk box.
Company A's internal numbers:
CID: User:
1000 -
Hi guys,
I am trying to register IP IAX2 phone to our
Asterisk server.
this is what I see on traffic debug between the
asterisk server and IP phone.
I do not see anything in asterisk
console.
Can somebody give me hints what could be the reason
that phone is not registering?
Thank you in
Jesse Guardiani wrote:
Hello,
What is the best way to ignore a DID and not pick up the line?
I don't want to incur charges on the line (T1 PRI), so would
Hangup pick up the line, then hang up? Or can I use Hangup?
Use the Congestion application.
___
ADEGOKE ARUNA wrote:
I need your help
I have a sangoma A104D on my dell server; I got card status ok with no alarm
If I dialed the extension 6210006, it shows the output as stated below, but
there is no ringing from the pstn number nor the iax softphone am using on
my pc.
I will be glad
way to go Matt!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
RothSent: Thursday, March 02, 2006 11:51 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Lowering Server Load
All,Just a quick update on our
I was talking about the web managers posted, if they were free,gpl,gnu.
Some are commercial, but the first ones posted looked very nice and I think
they are free, but was asking to the poster.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
Oops. Yes, Actually I do have _X. I didn't copy and paste what I had, I
just typed it from memory. Thanks for the tip.
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Friday, March 03, 2006 9:48 AM
To: Asterisk Users Mailing List - Non-Commercial
Thank you , but the pstn subscriber am calling is not ringing at all
But I can here ringing from my own softphone from zap channel.
Thankx
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Friday, March 03, 2006 7:01 PM
To:
Hi All,
I'm stumped on a weird problem. I have an * server working fine for local
SIP phones and IAX2 connections. We just provisioned a second Ethernet
port to attach to a local SIP provider.
PSTN calls incoming work fine:
PSTN - SIP Provider - SIP - *
but outgoing calls are not. Call setup
Yep :) We were using that before we got the hardware cancelers in.
Aaron
Kevin P. Fleming wrote:
Aaron Daniel wrote:
Thanks :) When we were using Mark2 with aggressive suppression, we had
zero problems, but decided to go with the hardware canceler in our new
gateway since hardware's supposed
Hello all,
I am new to [EMAIL PROTECTED] and am having a strange issue with both my new
Grandstream HT-286 BT-101. The issue is as follows:
Example is with BT-101 (HT-286 shows same behavior)
1) Device registers to Asterisk
2) I can place a call via the BT-101 out my Zap or SIP provider
3)
Jim Van Meggelen wrote:
Let me run something that's been floating about in my noggin by
everyone:
Given that Asterisk does not make use of dual core CPUs or dual
processors...
Jim,
That statement bothered me, because we are running Asterisk on a
multi-processor system to help accomplish
Would QoS on a managed switch solve the ARP problem?
I'm not sure about QoS, because we haven't tried it, but my initial
feeling is probably not. We solved our problem by separating the
network segments completely, which provides us with better security as
well as the quality we required.
I say
We have an external FXO/FXS, and use Asterisk as a call router. We
want to use G723 for the actual phone calls, because we have limited bandwidth
on our return direction. This has been working fine so far.
However, now we want to set up Asterisk to handle PBX menues and accept
extentions.
Hi.
Im facing a really bad voice quality when a
make calls between a tdm user and a sip user.
Take a look at the following scenario:
sip-user asterisk TDM22B(fxo)
PABX
and
PABX my-tdm-extension
When the sip-user places a call to my-tdm-extension, the
For the archives record.
Original Message
Subject:Re: [Asterisk-Users] problem with incoming peer (cisco as5400)
Date: Fri, 03 Mar 2006 11:14:50 -0600
From: Miguel [EMAIL PROTECTED]
To: Ron McCarthy [EMAIL PROTECTED]
References: [EMAIL PROTECTED]
[EMAIL
Anyone have any luck RMAing a Sipura phone since the Cisco take over?
Sipura only has support via email or fax to end users and I haven't
gotten a response to either for over 2 months.
Sipura's policy was to handle RMAs through resellers. Since taking over,
Linksys appears to have maintained
I have had the same issue with Sipura as well, they gave me quite the
run around on 1 bad 2002 ATA and decided to just forget about it and
bought a new one to save time.
On 3/3/06, Michael Sampson [EMAIL PROTECTED] wrote:
Its good for us to post thing about different companies customer
service.
It is my understanding that when you hear echo the problem is on
the other end. So if a caller complains they hear echo that is
something you should be dealing with, but if you hear echo that is the
phone companies fault. Now with a normal phone, the phone company will
only echo cancel long
Its good for us to post thing about different companies customer
service. I feel one of the most important points when buying a product
is if the company is going to stand behind it. I had purchased some
sipuras that did not work but was lucky and able to send them back to
the store I
On Mar 3, 2006, at 9:48 AM, Anton Krall wrote:
Guys.
I have a te100p with unicall and an E1 and Im having problem with DTMF
tones
but the weird thing is, I only have problems sending the tones to
certain
phone numbers, anybody seen this behavior?
Asterisk shows on the console the dtmf
Your best bet is to just not use fax machines. They are outdated
technology. With email there is little reason to use fax machines
anymore. But for some reason people just feel the need to hang on to them.
A good solutions is to get a fax machine that supports fax to email. We
have a Brother
I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the sip.conf
entry, I have [EMAIL PROTECTED] and vmexten=*98.
The light on the snom 320 turns on when I have voicemail and the retrieve
button dials the correct extensions.
However, the light turns off immediately after making
I guess if I was going to do this I would either have a sip adapter at
each phone. Or have to * boxes. One is connected to the PRIs. Then
connected to that via an IAX2 trunk is another asterisk box that is full
of the 24 port FXO/FXS cards digium sells. You could expand this as much
as you
HELLO everyone
I am having two alcatel 4600 digital phone PBXs .. They are situated in two locations 15km apart.
I want users or extension in both PBXs to be able to dial and receive calls from each others through those 30 channels in the E1 ..
I have line of sight so i am planing to use a
Michael Sampson wrote:
Your best bet is to just not use fax machines. They are outdated
technology. With email there is little reason to use fax machines
anymore. But for some reason people just feel the need to hang on to them.
There are still many valid uses for fax. The technology is not
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hey all,
First of all, hello again! Been a while since I've posted to the
list, but I've been here lurking and watching ;-)
Anyway, I wanted to pose a general question to the list to see
if it turns up new suggestions for everyone/me.
What is your
In theory I would say I agree how ever in practice... I have a PBX
(Merlin Legend) that I am connected to via PRI (10 foot pre-fab'ed
cable) and I get intermittent echo on the voip side. There is nothing
in between * and the PBX...
sean
On Fri, 2006-03-03 at 13:42 -0600, Michael Sampson wrote:
Michael Sampson wrote:
Your best bet is to just not use fax machines. They are outdated
technology.
It is older technology, true... but certainly it's not useless
technology. Certainly there is nothing yet to replace it properly. And
I could argue this on a technological standpoint, and
I had the same problem, I just set the voicemail button on the phone to dial the voicemail extension, but you will still have the problem (at least on the 360) if the user uses the Soft buttons below the display to access the voicemail.
On 3/3/06, Nabeel Jafferali [EMAIL PROTECTED] wrote:
I am
Anyone selling CON-SNT-CP7970 ?
-Dan
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On Mar 3, 2006, at 11:42 AM, Michael Sampson wrote:
It is my understanding that when you hear echo the problem is on the other end. So if a caller complains they hear echo that is something you should be dealing with, but if you hear echo that is the phone companies fault. Now with a normal
I am now receiving
spontaneous restarts ofasterisk on my system, with no apparent rhyme or
reason. I am using version 1.2.4, with zaptel-1.2.3 (downgraded from
1.2.4. I downgraded after the system started restarting spontaneously,
this week. I upgraded last Friday). I see no indication of
On Mar 3, 2006, at 9:49 AM, David Thomas wrote:
I'm doing an install for a client with the following requirements.
- 1 Million minutes of outbound calling
Per what?
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On Mar 3, 2006, at 9:57 AM, Bartosz Jozwiak wrote:
Hi guys,
I am trying to register IP IAX2 phone to our Asterisk server.
this is what I see on traffic debug between the asterisk server and IP phone.
I do not see anything in asterisk console.
Can somebody give me hints what could be the
Hello
Looking the SIP debug we see a change in the SETUP
message from the Asterisk 1.0.x version to the 1.2.4.
In the 1.2.4 we notice this line:
a=fmtp:18 annexb=no
This line cause problems in our plattform (We think
our SIP - h323 gateway can't parse this line)
Why this line its
Youll want to check the docco
against the SetGroup and CheckGroup applications,
although I think these have been deprecated in favour of a variable
type approach now.
Regards,
Mark
-Original Message-
From: Marc Archer
[mailto:[EMAIL PROTECTED]
Sent: Friday, 3 March 2006
On Mar 3, 2006, at 11:35 AM, Tom Vile wrote:
I have had the same issue with Sipura as well, they gave me quite the
run around on 1 bad 2002 ATA and decided to just forget about it and
bought a new one to save time.
So they gave you shitty support and you bought more?
What are you a microsoft
http://grandstream.com/BETATEST/HT488_496_386/
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