Have canreinvite set for your internal extens.
You can also have canreinvite enabled by default for all and use one or more of
the 't','T','h','H','w','W' or 'L' options set in your dial commands which will
override the canreinvite option and not send re-invites.
cheers
- Ben
--- On Sat,
Hi Bilal,
Yes it is definitely possible. And I've done it myself for a couple of our
clients.
Does that answer your two questions?
cheers
- Ben.
--- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote:
From: bilal ghayyad [EMAIL PROTECTED]
Subject: [asterisk-users] AGI and prepaid
Look at the canreinvite option.
- Original Message
From: Rizwan Hisham [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 17, 2008 3:20:40 PM
Subject: [asterisk-users] dtmf passthru
hi all,
Is
From: Benjamin Jacob [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 17, 2008 4:45:13 PM
Subject: Re: [asterisk-users] dtmf passthru
Look at the canreinvite option.
- Original Message
From
Hello Roland,
You can use the cmd Read for this.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
Pretty straight forward. Whenever you need to accept DTMF input from the user
collect the required digits using Read; check the collected digits; if yes jump
to required extension; else
Use SendDTMF.
--- On Thu, 7/3/08, Neha Punia [EMAIL PROTECTED] wrote:
From: Neha Punia [EMAIL PROTECTED]
Subject: [asterisk-users] (no subject)
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Date: Thursday, July 3, 2008, 10:29 AM
Hi
I m making a call from one
Care to explain the scenario Vincent?
Is it a SIP peer?
what is the DTMF mode set? etc.
--- On Tue, 7/1/08, Vincent [EMAIL PROTECTED] wrote:
From: Vincent [EMAIL PROTECTED]
Subject: [asterisk-users] User unable to use DTMFs?
To: asterisk-users@lists.digium.com
Date: Tuesday, July 1,
If by realtime, you mean to be able to read the MOH class from a DB and set
MusicOnHold, then I think you should try func_odbc.
Have never tried it, but reading the workings of it, it seems to be possible to
achieve this.
Let me know if you succeed in it.
- Ben.
--- On Tue, 7/1/08, Nhadie
modprobe zaptel; modprobe ztdummy
That will start zaptel and ztdummy after the 'zaptel stop'. Then restart
asterisk.
--- On Wed, 7/2/08, Doug Crompton [EMAIL PROTECTED] wrote:
From: Doug Crompton [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Choppy audio
To: Asterisk Users Mailing
I think, you can use the 'h' extension to invoke scripts (DeadAGI to be more
precise) on hungup channels.
use something like this :
exten = _X., 1, NoOp(got a call)
exten = _X., n, Dial(somexten}
exten = h, 1, DeadAGI(hangupScript.sh)
--- On Fri, 6/13/08, Robor Oghene [EMAIL PROTECTED]
Hello ppl,
Is there anyway to control a call mid-way in terms of sending a re-INVITE with
say sendonly, etc. to mute one call leg of a bridged call ??
Looked around, so far, doesnt seem to be possible.
If it's not, I think it's quite an important feature (re-INVITES mid-session)
for a B2BUA.
ringing event
To: asterisk-users@lists.digium.com
Date: Thursday, May 8, 2008, 12:00 AM
On Thu, May 08, 2008 at 12:19:52AM +0300, Atis Lezdins
wrote:
On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Benjamin Jacob schrieb:
Anyway in Asterisk to update a DB
Hello ppl,
Are the channel names generated on 'Dial's supposed to be unique?
I see the channel names repeating on my asterisk box. I just wanted to confirm
this.
Can anyone point me to the lines of code where the channel name is
generated/calculated? I tried looking, but it looks like quite a
, May 15, 2008, 11:46 AM
Benjamin Jacob wrote:
Are the channel names generated on 'Dial's
supposed to be unique?
I see the channel names repeating on my asterisk box.
I just wanted to confirm this.
Can anyone point me to the lines of code where the
channel name is generated/calculated? I
Hello ppl,
Anyway in Asterisk to update a DB/ do some action on
events like ringing.
The issue is I need to be able to hangup/cancel a
call, if it's ringing(decided by the admin). This is
independant of the timeout that we can specify in the
Dial command.
If I could somehow update a DB with
Last I was working on it, it did indeed NOT look at
sip.conf with realtime architecture being used.
But why take chances anyway? Move all the relevant
conf files from /etc/asterisk to some other place to
be safe.
cheers
- Ben.
--- Rilawich Ango [EMAIL PROTECTED] wrote:
HI,
Does asterisk
Hiren,
Not really clear as to what are the things you exactly
want.
List them out clearly.
Before you do that, do google and read up on
Asterisk+IVR
Asterisk+agi
Your need to calculate sum of birthdate digits etc can
be achieved using AGI scripting.
cheers
- Ben.
--- Hiren Mistry [EMAIL
http://www.openvox.com.cn/products_detail.php?genre_id=9id=28
If you can get the bare card, you can use it for
timing with a little
magic that can be found via google. If not, get one
with an FXO or
FXS and you will add a little flexibility and have
real hardware
timing.
If you
-
In the process of cleaning up unnecesary
processes, I
came across this line :
/usr/sbin/vmware-guestd --background
/var/run/vmware-guestd.pid
GASP so does this mean this is a virtual
machine??
I have got no idea about virtualization yet. So
how do
I confirm if
OK, I think you need to home in on the differences
between the server(s)
that work fine and the one that doesn't.
As I said in my other mail, the faulty one is a
.. mono processor machine, with SMP turned on
.. running CentOS 5
.. with kernel : 2.6.18-53.1.13.el5
There are other kernels
Hello ppl,
One on my clients' machine had Asterisk 1.4.4. installed. The complained of
choppy Playback of gsm files.
So scouring the internet gave me the solution of installing ztdummy and loading
it as a module.
Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and
Are my messages getting through?
This is urgent!! Any pointers?
Benjamin Jacob [EMAIL PROTECTED] wrote: Date: Thu, 24 Apr 2008 23:23:08 -0700
(PDT)
From: Benjamin Jacob [EMAIL PROTECTED]
Subject: Playback / Background / Read choppy, but musiconhold fine, even with
ztdummy
To: asterisk-users
Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED],
Benjamin Jacob wrote:
One on my clients' machine had Asterisk 1.4.4. installed. The complained of
choppy Playback
of gsm files.
So scouring the internet gave me the solution of installing ztdummy and
loading
Benjamin Jacob [EMAIL PROTECTED] wrote:
Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED],
Benjamin Jacob wrote:
One on my clients' machine had Asterisk 1.4.4. installed. The complained of
choppy Playback
of gsm files.
So scouring the internet gave me the solution
bridge and I need to send DTMF (the
bridge PIN) to it after connection. But alas, the reinvite happens before the
D() is executed.
The SIP gateway is MySIPGateway at 204.aaa.bbb.ccc.
cheers
- Ben.
Steve Davies [EMAIL PROTECTED] wrote: 2008/4/22 Benjamin Jacob :
[snip]
So, my question : once
Hello ppl,
Is there any other way to detect states like Ringing on SIP channels on
Asterisk?
Nvlinedetect is one way, but it seems to have disappeared from the face of the
earth!
Any pointers or does anyone have the code for NV* features?
Thanks in advance
- Ben.
Hello ppl,
I am using the Astman API Originate command to initiate a call to a user. On
connect of the user, I dial another user to bridge the call between the two.
I am using the Async option with the Originate command, as I don't want to use
Astman proxy yet. Is there any way to invoke a
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow.
The issue is that I need to send DTMFs after dialing the user because
most of the users are behind PBXes
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow.
The issue is that I need to send DTMFs after dialing the user because most of
the users are behind PBXes
clear this time.
cheerz
- Ben.
Steve Davies [EMAIL PROTECTED] wrote: On 21/04/2008, Benjamin Jacob wrote:
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow
I am using Realtime in virtually all my projects. So far, I haven't had
any major issues. It saves a lot of headache for profile/dialplan
updates, at least for me!
So I say, GO!
- Ben
Olivier wrote:
Hi,
I'm working on a 500 seats Asterisk project.
I'm wondering whether or not I should
Thanks good ppl!
Doug wrote:
At 10:02 12/17/2007, mail-lists wrote:
Same here - Gafachi has been great. Decent rates, very stable and great
voice quality.
I use Gafachi.com http://Gafachi.com and have good quality with no
minimum requirements. Try them at www.gafachi.com
Hello ppl,
Am looking at some PSTN termination providers in US. If this question
has been repeated, please point me to the correct link, as I've tried
searching the archives but have been unsuccesful so far.
I have come across quite a few companies which provide the same, such as :
Is this getting through??
EMAIL DISCLAIMER : This email and any files transmitted with it are
confidential and intended solely for the use of the individual or entity to
whom they are addressed. Any unauthorised distribution or copying is strictly
prohibited. If you receive this
simultaneous calls??.. will this correctly ensure the last call
retrieved from such DB was indeed the last call received?
Patrick wrote:
On Wed, 2007-11-28 at 11:07 +0100, Eric Smith wrote:
What is the easiest (simplest) way to do this?
Store the dialed number in the Asterisk DB and
duhhh !!
Patrick wrote:
On Wed, 2007-11-28 at 17:08 +0530, Benjamin Jacob wrote:
simultaneous calls??.. will this correctly ensure the last call
retrieved from such DB was indeed the last call received?
Look at the subject. He said *dialled* number, not received :)
Regards,
Patrick
an FSM
built into my billing engine, maintaining call states, etc. That seems
to be quite a daunting task to be done in a short time.
Any ideas anyone?or any similar experiences, in terms of performance,
scalability, etc. w.r.t both AGI scripts and AstMan API?
TiA
- Benjamin Jacob.
EMAIL
The reason could be bad routing, IPs used by multiple devices.. n so on...
Edwin Kariuki wrote:
Hi,
I have a voip platform that has a SIP server where about 450 sipura
phones adaptors register. On two occassions some phones (which were
previously working) have refused to register with
1000, 1001, 1002 all using the same voicemailbox 5.
Now, when someone calls 1000, and leaves a voicemail, I want to store
the fact that this voicemail was meant for extension 1000.
Similarly for 1001 and so on.
Any ideas anyone?
TiA
- Benjamin Jacob.
EMAIL DISCLAIMER : This email
for UDP
tcpdump -nnXs 0 udp -i eth0 -w name.cap
Btw, a pcap file (created on a linux server using tcpdump) capturing the
RTP(udp) traffic opened up in wireshark, wireshark doesn't really
format(or recognize) the packets as RTP, unlike the capture done live
from a wireshark configured to
Steve Edwards wrote:
On Thu, 15 Nov 2007, Benjamin Jacob wrote:
well.. if nothings working.. try putting in debug lines urself in the
code.. say
use system calls to write some debugging data into some temporary file
in ur perl code.
I'm a big fan of
syslog(LOG_ERR, I
well.. if nothings working.. try putting in debug lines urself in the
code.. say
use system calls to write some debugging data into some temporary file
in ur perl code.
let us know..
Matt wrote:
[EMAIL PROTECTED] agi-bin]# /usr/bin/perl -v
This is perl, v5.8.5 built for
Hello Steve,
I think Ray was talking more like the following setup (do correct me if
I am wrong):
User A (SIPcallId1) --- Asterisk (SIPcallId2) -- User B
In this case, the INVITE SIP callId received by Asterisk from User A is
different to that sent in the INVITE to User B.
I can get User
Also, how do you acces the second SIP call ID from the dialplan? Any
simple way to do this?
Benjamin Jacob wrote:
Hello Steve,
I think Ray was talking more like the following setup (do correct me
if I am wrong):
User A (SIPcallId1) --- Asterisk (SIPcallId2) -- User B
In this case
Hello All,
I've got the following warning messages a couple of days back:
/chan_sip.c: Maximum retries exceeded on transmission SIPcallId for
seqno 1 (Critical Response).
/Have got the warnings repeatedly for one Callid. If maximum retries
have exceeded why should it give me those warnings
show us the output of ur top command
Pezhman Lali wrote:
Dear
I am using an asterisk 1.2.7.1 , with postgres
and safe_Asterisk, for running, asterisk.
but there is a problem,
after 2-3 hours after restarting any things, top
shows me, that, two asterisk, are now running, and one
of them, gets
a2billing so far seems to be quite comprehensive compared to the other
freeware asterisk-based billing solutions available out there.
We are building our own billing solution(due to the very peculiar
requirements, one of which is to bill the callee, rather than the
caller). We are achieving
safe_asetrisk bundled with the package, does increase the file limits in
quite a neat way, with some other good setups.
Edit MAXFILES or SYSMAXFILES as required.
Also, I've read posts online, advising not to use safe_asterisk. Any
experiences on this one, anyone?
cheers
- Ben.
Jay R. Ashworth
You The Man, Anselm. Thanks for the details.
Anselm Martin Hoffmeister wrote:
Am Dienstag, den 11.09.2007, 17:11 +0530 schrieb Benjamin Jacob:
Thanks Anselm. This does clears a few things for me.
Tho, I couldnt find the patterns you mentioned in the docs(do point me
to the location if you
/debug messages into
/var/log/messages.
Same like your situation, the messages is comment (;)
and even the logges are written to the
/var/log/messages, so why that is happening?
Did u find answer for that?
Regards
Bilal
--- Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello Bilal,
You have to do
C F, I have nat=yes set by default for all my extensions(with
canreinvite=no). And things work fine.
Bilal, about Asterisk sending packets to public/private :
Asterisk will send packets to the public IP advertised by the msg/recv
from address. It is the NAT's headache on the endpoints network
accessing the * console via ssh.
thanks for ur help.
- Benjamin Jacob.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
Jacob
Sent: 04 September 2007 12:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] stop
- Benjamin Jacob.
EMAIL DISCLAIMER : This email and any files transmitted with it are
confidential and intended solely for the use of the individual or entity to
whom they are addressed. Any unauthorised distribution or copying is strictly
prohibited. If you receive this transmission in error
Hello good ppl,
Any way of stopping asterisk writing into syslogs or any other file, if
I set verbose 6 on the console?
All I want is the verbose output only on the console, nowhere else.
My logger.conf says :
console= notice,error
;messages = notice,warning,error
Thanks in advance.
- Benjamin
Btw, even the syslog line in logger.conf is commented :
; syslog.local0 = notice,warning,error
Benjamin Jacob wrote:
Hello good ppl,
Any way of stopping asterisk writing into syslogs or any other file, if
I set verbose 6 on the console?
All I want is the verbose output only on the console
Exactly the same lines as on the console.
Adrian Marsh wrote:
What logs are coming out to /var/log/messages?
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
Jacob
Sent: 04 September 2007 07:58
To: Asterisk Users Mailing List - Non
Here it is :
SIP01*CLI logger show channels
Channel Type StatusConfiguration
--- ---
Console Enabled- Notice Error
Tzafrir Cohen wrote:
On Tue, Sep 04,
to confirm
identities of end subscribers.
All corrections and suggestions welcome.
- Ben
Benjamin Jacob wrote:
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set(CALLERID(all) above supposed to change the
display name as above(Ben Jacob) and change the From URI
identities of end subscribers.
All corrections and suggestions welcome.
- Ben
Benjamin Jacob wrote:
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set
Anthony Francis wrote:
Hello good ppl,
A couple of questions for multiple pbxes
1. Is it possible to support multiple pbxes in one Asterisk box(using
contexts, etc.)?
2. Can we use the domain field in sip.conf to specify the different
domains for sip users, having one domain for each pbx?
I just
Hello good ppl,
A couple of questions for multiple pbxes
1. Is it possible to support multiple pbxes in one Asterisk box(using
contexts, etc.)?
2. Can we use the domain field in sip.conf to specify the different
domains for sip users, having one domain for each pbx?
I just tried registering two
anyone , to my query(abt multiple pbxes)? Apologies if I
am missing something elementary here.
cheerz
- Ben.
C F wrote:
Can you please get rid of your awfull long nonsense disclaimer?
On 8/1/07, Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello good ppl,
A couple of questions for multiple pbxes
Hello ppl,
Searched all over, but couldn't find anything conclusive.
Does an off-the-shelf version of Asterisk run without any issues on a
64-bit machine?
Does anyone have any 'conclusive' figures?
Apologies if this is a repeat question. Would appreciate if I could be
redirected to the
Pierre,
Thats exactly what Joanna said in her reply.
Check the client DTMF settings on your phones.
set it to rfc2833 or out-of-band, whatever the config says.
Grandstream by default have inband DTMF set, and usualy ulaw is
supported as well, and thats the reason ur grandstream works but
rfc2833 is the prefered way, as inband will work perfectly only with the
ulaw codec.
Pierre Marceau wrote:
Okay, in the SPA-941 admin I changed:
;DTMF Tx Method: Auto
DTMF Tx Method: Inband
and now it works.
Thanks!
Pierre
[EMAIL PROTECTED] 2/21/2007 12:09 AM
Pierre,
Thats
Make it
Goto(s-${DIALSTATUS})
cheerz
- Ben.
Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial
attempts in the s extension. Goto() is used in examples. Is the
prefix s- mandatory? Is it related to the original extension s?
(Apparently Goto(${DIALSTATUS})
If you already havent seen this:
http://dir.indiamart.com/impcat/video-telephone.html
cheerz
- Ben.
Crazy Boy wrote:
Hi Friends,
This is Chandra from India. I have installed and configured Asterisk
in our company. I want to provide Polycom IP 501 model phones to our
employees. I am unable
Davida,
You would also want to look at canreinvite option in sip.conf
http://www.voip-info.org/wiki-Asterisk+SIP+canreinvite
cheerz
- Ben.
Eric ManxPower Wieling wrote:
David Alcott wrote:
Is there a way to configure the Asterisk so that the RTP goes
directly between the Endpoints as
And ofcourz, be careful, with your fingers on the CLI or elsewhere, esp
on a production server.
cheerz
- Ben.
Benny Amorsen wrote:
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG When I exited the CLI and re-entered and pressed ctrl-c,
That's where your problem is. Use
Use astdb for such apps. Look at Lookupblacklist, similarly, you can
set up ur whitelist
http://www.asteriskguru.com/tutorials/lookupblacklist.html
Vernier Umali wrote:
I looked at the ex-girlfriend option and it's just part of what I
needed. What I do want is to setup a whitelist or
On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote:
What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter,
the last command in the history always defaults to 'stop now'. This is very
bad, and it's caused accidental shutdowns more than once.
thats prety
: Re: [asterisk-users] mwi for voicemail not showing up for
realtimeconfig.
Since I started using 1.4 I'm also not getting MWI. I am not using
realtime.
MARK.
Benjamin Jacob wrote:
Hello ppl,
Am using realtime odbc storage for voicemail, sip users/peers, static
for extensions and so on.
My
Hello ppl,
Am using realtime odbc storage for voicemail, sip users/peers, static
for extensions and so on.
My issue is I am not getting MWI for any fones, even tho I've got
rtcachefriends=yes in sip.conf
WIth tcpdump, I always see the NOTIFY going as
Messages-Waiting:.no
Hello ppl,
The scenario :
I restart asterisk, sip show peers shows nothing.
I make a call from 7013 to 7011.
I get the following o/p :
SIP Seeding peer from astdb: '7013' at [EMAIL PROTECTED] for 3600
SIP Seeding peer from astdb: '7011' at [EMAIL PROTECTED] for 240
And then the call goes thru.
AFAIK, ODBC helps you have any DB underneath, be it MySQL, PGSQL, etc.,
so why not go ahead with it?
cheerz
- Ben.
Norbert Zawodsky wrote:
Hi Peder,
I asked the same question some time ago.
Never got any answer... :-(
Norbert
Peder @ NetworkOblivion schrieb:
Is the storage of actual
Hello ppl,
Is it possible to send a REDIRECT from an Asterisk box, to an incoming
call??
e.g. A calling B, via Asterisk,
Asterisk sends redirect to A to contact C.
cheerz
- Ben.
___
--Bandwidth and Colocation provided by Easynews.com --
a lil bit of googling wud have answered you Tim.
Put in some effort next time anyway, for now :
http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+openhours
Wildheart wrote:
Hi,
I want to change my voicemail message based on the time of day. I would
like a message that says
Hello ppl,
Reading all over the net. Learnt quite a lot, but that has left me
confused-a-lot as well.
Need answers to a few questions. Before that, I have an ISP(fax gateway)
which will help me send/recv faxes using the T.38 protocol. I am using
Asterisk 1.2.12.1.
Now to the few questions I
Your offnet calls will be more than 4 digits, so use that to ur advantage.
so, for internal calls,
exten = _,1,Set(CALLERID(all)=Name ${CALLERID(num)})
or if u dont want to change the CLID at all.. dont do anything..
exten = _,1,NoOp(nothing)
else, for all external calls(4 digits)
exten
Just the answer I expected.
But, how do I get the IPs of the two parties?
Jon Schøpzinsky wrote:
You can use the CDR(userfield) value, to save the ip's in the CDR record.
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Benjamin Jacob
Sendt: 3
Hello ppl,
Any way to store the origination or termination IP addresses in CDRs?
cheerz
- Ben.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I know, I know, the wiki link for that one.
But wot I wanted were actual figures related to Asterisk n QoS.
How does Asterisk actualy handle and fare at the following QoS issues :
1) Delay
2) Jitter
3) Packet loss
These and more ideas are welcome.
cheerz
- Ben.
Not too sure, if this msg did reach the group, so resending.
---BeginMessage---
I know, I know, the wiki link for that one.
But wot I wanted were actual figures related to Asterisk n QoS.
How does Asterisk actualy handle and fare at the following QoS issues :
1) Delay
2) Jitter
3) Packet
Martin Joseph wrote:
On 2006-10-25 08:14:43 -0700, Noah Miller
[EMAIL PROTECTED] said:
Hi Matt -
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
On the customer's end I have the
Arnd Vehling wrote:
Jordan Novak wrote:
Has anyone created a GUI for this.
I am not sure what youre looking for but we developed a Voicemail
Manager:
= http://sip-syndication.com
best regards,
Arnd
Hello Vehling,
This product of yours, does it manipulate, files on the Asterisk
into
sipusers = odbc,mysql2,sip_buddies
sippeers = odbc,mysql2,sip_buddies
And realtime load sipusers username 1006 now returns data :-)
greets
Tijl Van den Broeck
On 10/23/06, Benjamin Jacob [EMAIL PROTECTED] wrote:
Make additional checks :
1) ensure u've unixodbc, unixodbc-devel installed, use
Any more ideas, esp from guys whove used this in their setp?
Benjamin Jacob wrote:
Giovanni,
Appreciate your lines mate.
But, Ive already read those, all over the net.
my qs inline :
amaflags : Categorization for CDR records. Choices are default, omit,
billing, documentation and choices
|
This is the output of the realtime load command:
realtime load sipusers name pippo
No rows found matching search criteria.
Thank's
Maury
- Original Message -
From: Benjamin Jacob [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Avi Miller wrote:
On 23/10/2006, at 2:35 PM, Eric ManxPower Wieling wrote:
Works for me. 1.2.12.1 with FreePBX. When I press *, I get a
password prompt. Entering my password gets me into the main
voicemail menu.
FreePBX is NOT Asterisk.
Yes, I know that. Hence the 1.2.12.1 *with*
Maurizio Pederneschi wrote:
Hi,
i have implemented Asterisk Realtime architecture with Odbc and MySql
DB. I have followed all the step of the documentation I found on the
Internet.
On the CLI, if I make odbc show I see that the DB connection is
UP, but if I make realtime load family
Hello ppl,
Can someone explain to me the meaning and use of the variables
accountcode and amaflags in sip.conf,etc.
Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. I
know, they are billing related, but not much beyond that.
Any ideas?
cheerz
- Ben.
..
Cheers,
Giovanni
2006/10/19, Benjamin Jacob [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Hello ppl,
Can someone explain to me the meaning and use of the variables
accountcode and amaflags in sip.conf,etc.
Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. I
Eric ManxPower Wieling wrote:
Benjamin Jacob wrote:
Hello ppl,
This post is to do with the variables 'nat' or 'canreinvite' for sip
entities.
Idealy users, wont be static, they could be roaming all over the
globe. So, setting someone as behind NAT, and disabling canreinvite,
etc
Conrad Wood wrote:
On Wed, 2006-10-18 at 13:24 +0200, Tzafrir Cohen wrote:
On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote:
On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote:
On Wednesday 18 October 2006 05:47, Conrad Wood wrote:
To do something
Tzafrir Cohen wrote:
On Wed, Oct 18, 2006 at 05:26:49PM +0530, Benjamin Jacob wrote:
Conrad Wood wrote:
On Wed, 2006-10-18 at 13:24 +0200, Tzafrir Cohen wrote:
On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote:
On Wed, 2006-10-18 at 08:55 +1300
Hello ppl,
This post is to do with the variables 'nat' or 'canreinvite' for sip
entities.
Idealy users, wont be static, they could be roaming all over the globe.
So, setting someone as behind NAT, and disabling canreinvite, etc.,
restricts the roaming capabilities of a user.
Is there any way
Hello ppl,
Is it possible to send SIP messages as response to the calling UA on
failure, for e.g. if a number is blacklisted I would want to send
Forbidden to the caller, not just for user comfort but also for testing
purposes?
I see only Congestion available which sends Service Unavailable.
Magnusson wrote:
Benjamin Jacob wrote:
Hello ppl,
Is it possible to send SIP messages as response to the calling UA on
failure, for e.g. if a number is blacklisted I would want to send
Forbidden to the caller, not just for user comfort but also for
testing purposes?
I see only Congestion available
On Tuesday 17 October 2006 10:31, Time Bandit wrote:
The one that never did a mistake, never did anything
so the q is.. will you be doing something a lot?? ;-)
... just kidding mate.. but thats a good line neway.
cheerz
___
--Bandwidth
1 - 100 of 139 matches
Mail list logo
