Hi,
Was just wondering about the bug fix in asterisk-1.6 Beta Release..
Its been said that Added an 'n' option to SpeechBackground to request that the
channel not get answered.
Can anyone brief me about what exactly this bug are they talking about. We had
issues with Background syntax working
I assume span 2 is set ti T1...
Also you should be using a crossover if your going from the card direct
to the channel bank. Remember an RJ48 crossover and an RH45 crossover
are not the same.. It you are using an RJ48 crossover and your span 2 is
T1 then try auto T1
If All else fails you can
Have you tried setting the card as being T1 instead of E1 for the port
connected to the channel bank?
PaulH
On Tue, 2008-03-25 at 15:33 +1100, Lee, John (Sydney) wrote:
Any luck with the channel bank?
Thanks for the reminder Paul but so far no luck.
I have been getting:
1) ***
I assume span 2 is set ti T1...
Thanks James. I will check.
Also you should be using a crossover if your going from the card
direct
to the channel bank. Remember an RJ48 crossover and an RH45 crossover
are not the same.. It you are using an RJ48 crossover and your span 2
is
T1 then try
Have you tried setting the card as being T1 instead of E1 for the port
connected to the channel bank?
Thanks Paul.
I was thinking about the same thing as I was leaving work today.
I will try to set the jumper just on port 2 and let you know.
___
--
Hi,
Nobody has clues on it?
Regards,
--
Sylvain Desbureaux
+33 296 051 380
-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED]
Envoyé : vendredi 21 mars 2008 17:08
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] Problem
Steve - Where are you buying your HPDL380 's ? I may need quite a # of
these for a new project.
What factors were the Most Important to you in selecting this product ?
What, if anything, is there any you do NOT like about these boxes ?
How many have you deployed ?
What is the largest box
Well it greatly depends on your budget. CDW or PCConnection are my
usual providers of server hardware. I like PCConnection, because I
have a very good relationship with my sales rep and he will often drop
VAR pricing further without being asked or with a small hint about the
price. In a pinch,
On 20/03/2008, Loic Didelot [EMAIL PROTECTED] wrote:
Hello,
I am having some troubles with Snom phones and maybe someone can help
me.
Let me say this: BLF and pickup works great with Polycomes and
Grandstream etc... So I think my problem might not be Asterisk related
but I am not 100%
What version of asterisk ?
Paste your errors in cli
Paste your extension.conf , only the part/context where sip client extension
is.
Paste output of 'sip show peers' in cli.
2008/3/25, Sneha Murganoor [EMAIL PROTECTED]:
Hi,
I have set up the asterisk in Fedora 7 with SJphone, SJphone is
Alternatively...
On 24/03/2008, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Monday 24 March 2008 04:02, mark morreny wrote:
Dear friends,
I am having problem with running a sample php and I can't figure out why.
I can run the sample.php using CLI but when I run it inside the dialplan
1.2 has no menuselect
1.4:
./configure
make menuselect (and you get into it after this command automatically)
2008/3/25, Rob Hillis [EMAIL PROTECTED]:
Only if you're trying to compile Asterisk 1.2. Asterisk 1.4 also has the
menuselect configuration, though for most applications you don't
Yes, it is that silly space!!! Thanks for all your help.
Thanks,
Mark
On Mon, Mar 24, 2008 at 5:02 PM, mark morreny [EMAIL PROTECTED] wrote:
Dear friends,
I am having problem with running a sample php and I can't figure out why.
I can run the sample.php using CLI but when I run it inside
How can I force soft hangup (if that makes sense)?
show channels reveals a stale sip channel. It's of
an analog phone behind a Grandstream ATA which was
communicating with another SIP softphone. The latter
crashed. A soft hangup of the softphone seems to have
worked but it doesn't for the
On Tue, 25 Mar 2008, Vieri wrote:
How can I force soft hangup (if that makes sense)?
show channels reveals a stale sip channel. It's of
an analog phone behind a Grandstream ATA which was
communicating with another SIP softphone. The latter
crashed. A soft hangup of the softphone seems to
On 25/03/2008, Gordon Henderson [EMAIL PROTECTED] wrote:
On Tue, 25 Mar 2008, Vieri wrote:
How can I force soft hangup (if that makes sense)?
show channels reveals a stale sip channel. It's of
an analog phone behind a Grandstream ATA which was
communicating with another SIP
Hello,
How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough
variables in (within) my custom Asterisk application?
I can't use chan_sip.c internal structures (such as sip_pvt) in my custom
application, because there's no chan_sip.h and I can't include it into my
Sorry for the hijack, but I was wondering if I could be pointed at how
to get BLF and pickup working with Polycoms? I can use their Buddy
option to get a basic BLF system working, but that does not understand
the ringing state, and does not allow call pickup.
This is what I do (it works most
Like many users I get my voicemails emailed to me, AND left on the asterisk
server, so that I can retrieve them by phone or by email. However, I was
frustrated that after I deleted a message in outlook that I still had to
delete it from asterisk manually.
So, I wrote a script that runs on the
On Tuesday 25 March 2008 03:34:44 [EMAIL PROTECTED] wrote:
Hi,
Nobody has clues on it?
Hi.
Apparently you missed my previous reply. Please see the archives:
http://lists.digium.com/pipermail/asterisk-users/2008-March/208236.html
--
Tilghman
___
Okay, another Cisco related issue (sorry!).
Single Asterisk box at location 1.
Single Cisco box at location 2, however the Cisco is also the PBX for
location 3 (same physical machine, calls routed via VoIP).
Trying to have Asterisk be able to call EITHER Call Manager location. The
single SIP
Hi,
anyone out there with the same problems and a possible solution to the
following?
The functions callparking and hold use the same transferdigittimeout in
features.conf.
While I think 3 to 5 seconds are enough to let the user find their keys on
the phone,
the double ammount of time ( 2 x 5
on an 8 pin connector (rj48) copper up facing away pins labled left to
right 1-8
side a 1 white/blue
side a 2 blue/white
side a 4 white/orange
side a 5 orange/white
side b 1 white/orange
side b 2 orange/white
side b 4 white/blue
side b 5 blue/white
Lee, John (Sydney) wrote:
I assume span 2 is
In a CallManager environment (currently 4.0, moving to 6.1 in the next
few months), can Asterisk completely replace Unity
as a voicemail system?
What works and what doesn't? MWI? Call Handlers? Does everything
work via a SIP trunk? Who has done this
and is willing to contact me?
Thanks.
Hi,
I'm very sorry I didn't see you answer.
I've put this and now the peer stays.
Thanks !
--
Sylvain Desbureaux
+33 296 051 380
-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Tilghman Lesher
Envoyé : mardi 25 mars 2008 14:28
À : Asterisk Users
On Mon, Mar 24, 2008 at 10:09:34AM -0800, Mojo with Horan Company, LLC wrote:
Distinctive Ringing might be available from your telecom provider.
mark morreny wrote:
Hi all,
I am using Digium PCI board to receive PSTN call through regular phone
line. It is no problem for me to
--- Steve Davies [EMAIL PROTECTED] wrote:
On 25/03/2008, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Tue, 25 Mar 2008, Vieri wrote:
How can I force soft hangup (if that makes
sense)?
show channels reveals a stale sip channel.
It's of
an analog phone behind a Grandstream
--- Steve Davies [EMAIL PROTECTED] wrote:
Using rtptimeout and rtpholdtimeout settings in
sip.conf
I set
rtptimeout=10
rtpholdtimeout=30
(just for testing; I know these values are way too
low)
then did a
CLI sip reload
and waited more than 30 seconds.
The SIP channel is still there (InUse).
Hi all,
can anybody tell me how to make auto-congest time configurable(different)
for every sip peer. I mean if i want to dial a local number then i should be
able to set the autocongest time to 15000 mili seconds, but if i dial an
international number then i should be able to set the auto-congest
On Tuesday 25 March 2008 07:51:13 Mindaugas Kezys wrote:
How can I get peerip, recvip, from, uri, useragent, peername,
t38passthrough variables in (within) my custom Asterisk application?
I can't use chan_sip.c internal structures (such as sip_pvt) in my custom
application, because there's no
On Tuesday 25 March 2008 10:17:54 Vieri wrote:
--- Steve Davies [EMAIL PROTECTED] wrote:
Using rtptimeout and rtpholdtimeout settings in
sip.conf
I set
rtptimeout=10
rtpholdtimeout=30
(just for testing; I know these values are way too
low)
then did a
CLI sip reload
and waited more
Has anyone put together a patch which would allow a different
fromstring and emailbody based on context? Or any other way to have
more than one fromstring and emailbody per server?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On Tue, 25 Mar 2008 14:58 +0200, Yehavi Bourvine +972-8-9489444
[EMAIL PROTECTED] wrote:
[snip]
LOL. Very creative :) Thank you for the suggestion. I can work with that!
Steve
___
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I really have to learn to do more playing around before asking the group
these questions!
Here's the solution:
Create a second SIP Security Profile (but not on port 5061, since it's
reserved for secured connections...I used 5080).
Create a second trunk port using this second SIP security
On 25/03/2008, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Tuesday 25 March 2008 10:17:54 Vieri wrote:
--- Steve Davies [EMAIL PROTECTED] wrote:
Using rtptimeout and rtpholdtimeout settings in
sip.conf
I set
rtptimeout=10
rtpholdtimeout=30
(just for testing; I know these
Aaron wrote:
Okay, another Cisco related issue (sorry!).
Single Asterisk box at location 1.
Single Cisco box at location 2, however the Cisco is
also the PBX for location 3 (same physical machine, calls
routed via VoIP).
Trying to have Asterisk be able to call EITHER Call Manager
25 mar 2008 kl. 16.54 skrev Chris Carey:
Has anyone put together a patch which would allow a different
fromstring and emailbody based on context? Or any other way to have
more than one fromstring and emailbody per server?
Minivoicemail has limited voicemail capabilities, but has
multiple
HI,
We need to get our number into the White Pages.
Has anyone here actually tried it?
Thanks,
--
Leonard Burton, N9URK
http://www.jiffyslides.com
[EMAIL PROTECTED]
[EMAIL PROTECTED]
The prolonged evacuation would have dramatically affected the
survivability of the occupants.
--- Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Tuesday 25 March 2008 10:17:54 Vieri wrote:
--- Steve Davies [EMAIL PROTECTED] wrote:
Using rtptimeout and rtpholdtimeout settings in
sip.conf
I set
rtptimeout=10
rtpholdtimeout=30
(just for testing; I know these values are way
Hi Kevin I've just arrived from my holidays I have reviewed my emails and saw
that for some reason most part of my last message appears to be cut off.
Continuing with the previous discussion, can you provide an example skeleton
code of how the new registration way works on asterisk 1.4 and
jonas boering wrote:
Hi Kevin I've just arrived from my holidays I have reviewed my emails
and saw that for some reason most part of my last message appears to be
cut off.
Continuing with the previous discussion, can you provide an example
skeleton code of how the new registration way
Hi list
I plan to connect two asterisk box using libss7 ,i read the list messages (
thanks for this great jop) , i installed all the packegs with digium single E1
link in both boxes with cenos 5 and every thing is looking ok excact when i am
trying to call using sip channel it shows some
Hi,
I am having a strange problem with attempting to get voicemail-to-mysql to
work.
The biggest problem is that I am not able to store voicemail into database.
So, I followed the
instructor found on the web:
Updated the /usr/src/asterisk/apps/Makefile to have
USE_MYSQL_VM_INTERFACE=1 and
--- Tilghman Lesher
[EMAIL PROTECTED] wrote:
The SIP channel is still there (InUse).
Channel Location State
Application(Data)
SIP/6010-b38d53e0[EMAIL PROTECTED]:8 Up
Dial(SIP/4053||tTwW)
Should I interpret the above that it's in an
infinite
Thank you!
You saved my day!
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tilghman Lesher
Sent: Tuesday, March 25, 2008 5:43 PM
To: Asterisk Users Mailing List - Non-Commercial
We need to get our number into the White Pages.
Has anyone here actually tried it?
You probably need to be more specific about which country you're in so people
can help you.
If you're in the UK, I can confirm we have plenty of customers with numbers in
the Yellow Pages, as well as the BT
On Tue, Mar 25, 2008 at 11:19 AM, Jiffy Slides Leonard Burton
[EMAIL PROTECTED] wrote:
HI,
We need to get our number into the White Pages.
Has anyone here actually tried it?
It's not just Voip numbers. We've got a PRI from XO that (even though
they say otherwise) we can't white pages
Hello,
My provider has told me that I need to change the RTP payload time to 10
milliseconds instead of the default of 20 to make their RTP packet
communication work. Is there anyway to do this in Asterisk?
Also, what does RTP payload mean and why would it effect the sending of RTP
packets?
On March 25, 2008 02:15:42 pm Lacy Moore wrote:
I think that is one of the biggest things that businesses overlook
when switching to Voip. It's hard to get in the directories.
I have to say that it's been many years (well before voip) that I've gone to
the directories. Google and yellow
Dear all,
I am able to send and receive fax with Asterisk + iaxmodel + hylafax. What
I want to be able to do is to
1. Stored the received fax in mysql
2. Send an email notification to he user corresponding to the incoming phone
number
3. Send a SMS notification to the user's mobile phone
In the
Lacy Moore wrote:
On Tue, Mar 25, 2008 at 11:19 AM, Jiffy Slides Leonard Burton
[EMAIL PROTECTED] wrote:
HI,
We need to get our number into the White Pages.
Has anyone here actually tried it?
It's not just Voip numbers. We've got a PRI from XO that (even though
they say
If you are running 1.4, check rtp-packetization.txt under
doc/ directory from source distribution.
Cheers,
--
exvito
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
mark morreny escribió:
Dear all,
I am able to send and receive fax with Asterisk + iaxmodel + hylafax.
What I want to be able to do is to
1. Stored the received fax in mysql
2. Send an email notification to he user corresponding to the incoming
phone number
3. Send a SMS notification to
Elliot Murdock wrote:
Hello,
My provider has told me that I need to change the RTP payload time to
10 milliseconds instead of the default of 20 to make their RTP packet
communication work. Is there anyway to do this in Asterisk?
Sure, in sip.conf just define the codecs like this:
On Tue, Mar 25, 2008 at 1:39 PM, Andrew Kohlsmith (lists)
[EMAIL PROTECTED] wrote:
On March 25, 2008 02:15:42 pm Lacy Moore wrote:
I think that is one of the biggest things that businesses overlook
when switching to Voip. It's hard to get in the directories.
I have to say that it's been
That's from asterisk-addons, you can ignore that error.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mark morreny
Sent: Tuesday, March 25, 2008 10:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Have problem
BUT - does it cure baldness and impotence?
Shouldn't anyone who posts stuff like this be banned? After flogging
tarring and feathering, of course!
Peg Leg O'Brien
BerkHolz, Steven wrote:
Asterisk work does not pay all of my bills, so I have joined up with a
company that has a very good
Hey List,
I'm trying to work out a resolution for domain based SIP authentication. We
are working on a virtual PBX type product and want to allow username
overlaps on separate domains so that [EMAIL PROTECTED] and
[EMAIL PROTECTED] are entirely separate of each other. I've found this
possible
Hi Mike,
Do you have any idea what is causing my voicemessages not being stored in
mysql?
Thanks,
Mark
On Wed, Mar 26, 2008 at 3:17 AM, Mike Fedyk [EMAIL PROTECTED] wrote:
That's from asterisk-addons, you can ignore that error.
-Original Message-
*From:* [EMAIL PROTECTED]
Vieri [EMAIL PROTECTED] writes:
Should I interpret the above that it's in an infinite
loop trying to dial/reach SIP/4053?
They are just stuck channels. It's simply a bug in 1.2.x. Fortunately
it's fixed in 1.4.x. We upgrade our customers to 1.4.x when they hit
that bug.
/Benny
After lots of interest I've stopped emailing people the script and have made
it available for download from www.generationd.com Look in the Downloads |
Asterisk section.
Be sure to read the readme AND the top of the script for instructions...
___
--
Does anyone have any idea what would cause distorted audio but ONLY for
DTMF tones coming in over our analog lines. (The analog interfaces are
X100P's). I have carefully adjusted the gains in the zapata.conf using
a local test line after trying various settings with no gain or just
random
If you want barebones where you add your own processor, RAM, hard drives, and
options, try SuperMicro brand servers. They are thousands of dollars less than
the big (fat) names like IBM and HP/Compaq, but very good quality.
I've built several clusters of computers with SuperMicro systems.
Guido Hecken wrote:
Hi,
anyone out there with the same problems and a possible solution to the
following?
The functions callparking and hold use the same transferdigittimeout in
features.conf.
While I think 3 to 5 seconds are enough to let the user find their keys on
the phone,
the
I've had good luck with these guys:
http://rackmountsetc.com/
supermicro have never failed me yet.
On Tue, Mar 25, 2008 at 6:03 PM, Jesse Molina [EMAIL PROTECTED] wrote:
If you want barebones where you add your own processor, RAM, hard drives,
and options, try SuperMicro brand servers.
Another useful script for those interested
On the www.generationd.com web site you will now find the asteriskcontrol
script file. This script can automatically restart Asterisk (gracefully)
following a change in external IP address - for dynamic IP hosts. As well,
it can update the
ok - but, who do you call for HW problem ? HP has all levels of
warranties all depending on how much $$
you want to spend. What do you do if you buy and install Supermicro ?
HP also has 24x7 support center , again not for free, what do you do
with Supermicro ???
I am really interested
Hi list
what do you think is the best way to replicate an asterisk mysql realtime
database ,
i'm setting up a cluster , and i obviously need to ensure the database high
availability
i've been reading about DRBD and the internal replication functionality of
mysql , but i dont really know if its
I was thinking about the same thing as I was leaving work today.
I will try to set the jumper just on port 2 and let you know.
Yes, that fixed the problem.
Thanks James and Paul.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Hi,
I am running asterisk 1.4.18.1 when I do make menuselect it makes it and the
last line is menuselect changes NOT saved! and then it goes back to the
prompt
On Tue, Mar 25, 2008 at 6:00 AM, Grygoriy Dobrovolskyy [EMAIL PROTECTED]
wrote:
1.2 has no menuselect
1.4:
./configure
make
Does it cure DUMB people who spost their garbage on asterisk users list?
2008/3/24, John Novack [EMAIL PROTECTED]:
BUT - does it cure baldness and impotence?
Shouldn't anyone who posts stuff like this be banned? After flogging
tarring and feathering, of course!
Peg Leg O'Brien
BerkHolz,
Okay,
I think I have found part of the problem. When I do make menuselect at the
end it reads Install ncurses to use menu interface!. I already have
ncurses and ncurse-devel installd so I am perplexed as to why this is coming
up. Any thoughts?
On Tue, Mar 25, 2008 at 9:06 PM, Kyle Gibbons [EMAIL
I am trying DB replication with MySQL. I have two nodes, and Linux-HA
running on both of them. The first idea is simply have the master
database in one server, where both nodes send the SQL queries and in
the other node a second DB replicated from the first one. In the case
of the master database
All,
Thank you very much for your help, I have solved the problem. After
installing ncurses-devel I had to completely delete the zaptel directory(I
know I was asking about Asterisk, but I was having the same problem and of
course was starting with zaptel install). I tried doing make dirclean, but
On Tuesday 25 March 2008 20:43:49 Edgar Guadamuz wrote:
I am trying DB replication with MySQL. I have two nodes, and Linux-HA
running on both of them. The first idea is simply have the master
database in one server, where both nodes send the SQL queries and in
the other node a second DB
Good to here,
I know the time off set US - AU is terrible when you need support.
Lee, John (Sydney) wrote:
I was thinking about the same thing as I was leaving work today.
I will try to set the jumper just on port 2 and let you know.
Yes, that fixed the problem.
Thanks James and Paul.
Does anyone have use for a broadcast/annouce app?
I wrote SystemAnnounce which will play a specified file to all active channels
(in an UP or bridged state). This was originally to tell users to get off the
system, but there are several other uses...
I also wrote a new CallPickup and CallPark
If you can't troubleshoot a hardware problem, then you should definitely not be
thinking about this. Going with a support-yourself plan is not for everyone,
especially if you don't have good hands-on hardware ability local to where the
systems are.
The cost savings can be significant enough
On Tue, 25 Mar 2008, Justin Newman wrote:
Does anyone have use for a broadcast/annouce app?
I wrote SystemAnnounce which will play a specified file to all active
channels (in an UP or bridged state). This was originally to tell users
to get off the system, but there are several other
Good to here,
I know the time off set US - AU is terrible when you need support.
I have continued to configure the analogue phone by just adding new
extensions (just like any VOIP phone) to extensions.conf as follows:
exten = 5162,1,SetMusicOnHold(cpwr)
exten = 5162,n,Dial(Zap/32,20)
exten =
what do you think is the best way to replicate an asterisk mysql realtime
database ,
i'm setting up a cluster , and i obviously need to ensure the database high
availability
i've been reading about DRBD and the internal replication functionality of
mysql , but i dont really know if its
I think you can set callerid's in zaptel.conf for each analog port - I
did that for a client a while ago. (from memory)
PaulH
On Wed, 2008-03-26 at 15:53 +1100, Lee, John (Sydney) wrote:
Good to here,
I know the time off set US - AU is terrible when you need support.
I have continued to
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