Ken, Im having problems with the time on my polycoms, it doesn't matter
which sntp server and offset I enter, the phone wont take the offset into
account, Ive tried entering it directly on the phone and also on the .cfg
file but no luck, any tips?
|-Original Message-
|From: [EMAIL
I spent a days or two on this and in the end did
Musiconhold.conf
[livestream1]
mode=custom
application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 -@
/etc/asterisk/stream.playlist
Then in stream.playlist I just put the links from Shoutcast I wanted to
use
Now thas confusing to me.. How do you actually take 16 calls at a time? I
see 301's have 2 line keys.. And each can handle 16 calls... How do you
actually take all 16 and switch between all of them?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
Speaking of rings... Is there any way to make the Polycoms have a default
ring volume after rebooting? Seems everytime I reboot mines they all go back
to a very low ring volume.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Bill Michaelson
|Sent:
On Thu, 2006-02-23 at 08:09 +, Lee Archer wrote:
I spent a days or two on this and in the end did
Musiconhold.conf
[livestream1]
mode=custom
application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 -@
/etc/asterisk/stream.playlist
Then in stream.playlist I just put the links
Hi there
I am having trouble trying to configure a RT31P2 to work on a local network
Using asterisk 2.4
Firmware on router 1.30
Device is unlocked
I just keeps coming up with Can not connect to server.
I would also assume it would almost use the same configuration as a pap2 if
anyone can
Is there a way to have extensions automatically created for
registered sip users ?
I did some investigation and found some hope in chan_sip with
relation to the somewhat undocumented usereqphone option but i may be
totally off track.
All i want to be able to do is send a call to [EMAIL
What does :
CUSTOM_1 se.rt.8.name=Custom 1 se.rt.8.type=ring
se.rt.8.ringer=5 se.rt.8.callWait=7 se.rt.8.mod=1/
Ringers = 5
Callwait = 7
And mod = 1 stand for?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Noah Miller
|Sent: Thursday,
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
CDW and other large resellers like them have a difficult time selling
service contracts. The issue is they _must_ provide Cisco with a serial
number (of the phone) which is checked by Cisco to see if the company
...
First they are
What are people doing with the minibrowser? Any nice stuff been done out
there?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Andrew Kohlsmith
|Sent: Friday, February 10, 2006 1:35 PM
|To: asterisk-users@lists.digium.com
|Subject: Re:
On Thu, 2006-02-23 at 16:46 +0800, Nathan Alberti wrote:
Is there a way to have extensions automatically created for
registered sip users ?
in sip.conf
regcontext=sipregistrations
that adds them to sipregistrations, you can make that anything you want
however I am willing to bet there might
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I installed FC4, ran command, # yum install asterisk. A bunch of stuff
happened, but can't locate .conf files. I have a list of files:
/usr/share/doc/asterisk-1.2.4/configs/features.conf.sample
On 23/02/2006, at 4:56 PM, trixter aka Bret McDanel wrote:
On Thu, 2006-02-23 at 16:46 +0800, Nathan Alberti wrote:
Is there a way to have extensions automatically created for
registered sip users ?
in sip.conf
regcontext=sipregistrations
that adds them to sipregistrations, you can make
Adam Robins ha scritto:
Thanks, but we already have the TOS bits set to 0xB8, which matches
the QoS settings in our switches and routers.
This is definitely something that changed in the 1.07 to 1.24
upgrade. We have a pair of identical 1.07 servers connected via the
same network pipe that
change context to context=remote in [general] in sip.conf
you missing registration of peer :)
turby
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of btb
Sent: Thursday, February 23, 2006 4:10 AM
To: Asterisk Non-Commercial Discussion Users Mailing List
[dial]
exten = _X.,1,Dial(SIP/${EXTEN})
exten = _X.,2,Congestion
exten = _X.,102,Busy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathan Alberti
Sent: Thursday, February 23, 2006 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
to register my Asterisk with a SIP provider I use the following
syntax, as shown in the default sip.conf:
register = 2345:[EMAIL PROTECTED]
where
[sip_proxy]
type=peer
context=from-messagenet
host=sip.messagenet.it
port=5061 - please note this one!!!
5061 is provider's port I
I am not a mySQL expert (obviously), my limited SQL experience is with
MS SQL where stored procedures and views are an option.
This is with mySQL 4.x, so no views.
I'm no an expert too, but even if the algorithm is right and seems to
bring some optimization I think mysql way of do
Chris Bagnall wrote:
Greetings all,
Has anyone managed to get dialplan status hints working across multiple
servers? I've separated a load of SIP users out across 2 servers today, but
it'd be useful if they could still see each others' status.
I've replaced the various hint lines for the sip
Isaac Xiao wrote:
We have the same issue happened to all Asterisk versions of 1.2.X (I
tried all). In CLI, it shows “-- Incoming call: Got SIP response 500
Internal Server Error back from 192.168.2.104”. Once you see this msg,
the buddy watch won’t work any more until you reboot the phone. I
Nathan Alberti wrote:
Is there a way to have extensions automatically created for registered
sip users ?
I did some investigation and found some hope in chan_sip with relation
to the somewhat undocumented usereqphone option but i may be totally
off track.
All i want to be able to do is
Johnathan Corgan wrote:
btb wrote:
[7508] ;ipkall
type = peer
dtmfmode = rfc2833
context = remote
callerid = ipkall incoming 7508
nat = no
You've configured this entry as a peer, which is for dialing out, versus
as a user, which is for incoming calls. Solution is to change to
'type=user'.
Hi,
I have several Polycom 501 connected to asterisk. The phone has an
ACD-login function that I'd like to use. But I can't find find much
information about this.
I've read a post on [EMAIL PROTECTED]
(http://bugs.digium.com/view.php?id=6119) about this function but I'm
not really clear on if
file sip.conf:
register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
example:
register = 531:[EMAIL PROTECTED]:5061/1234
file extensions.conf
exten = extension,1,1,Dial(number)
exten = extension,1,2,HangUp
example:
exten = 1234,1,1,Dial(SIP/1)
I would never recommend using a type=friend for a service provider
connection. You need one peer for calling out and another for receiving
calls, or at least add a host=hostname of provider's server to
enable matching on IP on incoming calls.
The problem here is, as you figured out
I am trying to put a Shorewall firewall in front of my PBX, all the
other port forwards work fine but forwarding port 4569 to the PBX is not
working, it is being logged as rejected even though there is a DNAT rule
in shorewall.
Anyone seen this and have a solution?
Are you sure its
We could implement this in SIP, by forcing an outbound
subscription, but if all the servers are Asterisk servers
there has to be more simple ways to solve this as well as
cross-server voicemail notification.
Could you elaborate on that please? I'm almost certain to come across the
Hi there,
I have bought 70 sipura 841 phones for a customer of
mine.
When following the mass provisioning guide in the
admin manual for the sipura, I see it download the spa841.cfg file from my tftp
server
Sometimes the phone also downloads is phone specific
file via tftp, and it
I use Asterisk with a HFC-S ISDN BRI card.
This card needs bristuff patch from Junghanns.net.
After upgrading to Asterisk 1.2.x, my CDR reports (located in
/var/log/asterisk/cdr-csv/Master.csv*) are broken.
Instead of telephone numbers, I get random characters like 'H? or $%.
Sometimes,
On 02/22/06 11:08 C F said the following:
http://bugs.digium.com/view.php?id=5574
That is a patch that will do just that.
while an app is nice, followme could have been done thru some nifty
dialplan work as well.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL
On 02/22/06 23:11 Roger Lewau said the following:
Connected to Asterisk 1.2.4 currently running on ns2 (pid = 47562)
Verbosity is at least 9
-- Remote UNIX connection
-- Executing VoiceMailMain(SIP/asterisk-0946, @sip) in new stack
-- Playing 'vm-login' (language 'se')
--
Hi,
we've proof of conecpt system for speech recognition on Asterisk. We would
like to send results of recognition back to user in standard way.
Currently we're considering using sendtext command and it works with
Firefly. But I'm curious what soft or hard ip phones that can connect to
On 2/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I have several Polycom 501 connected to asterisk. The phone has an
ACD-login function that I'd like to use. But I can't find find much
information about this.
I've read a post on [EMAIL PROTECTED]
Hi,
example:
register = 531:[EMAIL PROTECTED]:5061/1234
Unfortunately this doesn't really fit my needs.
/1234 means [EMAIL PROTECTED], where default is the context specified in
the general section of sip.conf:
[general]
context=default ; Default context for incoming
Thanks!
Do you have any suggestions on what I might do next. I have the phones, I have
asterisk, and I have everything setup. But i can't get the login to work with
the Polycom function. Nothing happens...and I can't find any readmes' or
manuals.
Regards,
Jan
-Ursprungligt
Thanks,
We already have a cron reboot of all of our Asterisk servers every
night. We've been doing this for over a year due to memory leak issues.
After 2 weeks of messing around with every conceivable IAX2 and
jitterbuffer configuration, I switched to SIP yesterday. Complaints
went from 10-20
After rebooting, I can make one outgoing call successfully. Subsequent
calls don't work - the 360 just seems to do nothing after pressing the
OK button (but I can cancel the call, the phone isn't frozen). The
Asterisk console shows the first call going through, but nothing
appears for the
Has anyone been able to get the IP501 to discover the FTP
server IP address (via dhcp or dns) and download 100% of the config from a
provisioning server?
We are still having to touch each unit to enter the ftp
server address and password, as well as set many of the options that will not
Damon,
I have no problem provisioning 501s
through tftp. The tftp address is distributed via dhcp.
Thx,
Wojtek
From: Damon Estep
[mailto:[EMAIL PROTECTED]
Sent: Thursday, February 23, 2006
8:09 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] auto
provision
Would you mind sending a sample config
file?
Are you able to set the passwords (user
and admin)?
Any ideas on ftp or https vs. tftp for
better security?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wojciech Tryc
Sent: Thursday, February 23, 2006
6:15
I've having a big problem after having upgraded to 1.2.4 and
chan_capi-cm 0.6.4
When making outgoing calls I don't seem to have any control over the CLI
that is presented to the called party -- it can be any one of the MSNs
allocated to the line, allocated on what seems to be a random basis.
Been using mpg123 for moh for the last two years or so. However, when
I have * config errors, often times get a endless stream of console
messages and need to kill the two mpg123 processes.
Is there an alternative to mpg123 that eliminates that issue?
I see references in musiconhold.conf
Hi!
I'm trying to get the leds working on my snom phones (320/360).
Hinting rules are added to the extensions.conf, the function keys of the
phones are programmed. When I start the asterisk and then the phones, I
see the following:
asterisk*CLI show hints
-= Registered Asterisk Dial Plan
Alexander, Perhaps I'm wrong, but I have a server here next to my desk
(IBM e325) and I tried to fit a normal pci card into it. The slots are
completely different and the card would not fit.. this was just a pci
dvi video card. The server specs say that it is using PCI-X technology
for the slots
Conrad Wood wrote:
Does the patch add any functionality to atftp that tftpd-hpa[1] doesn't
have?
This patch adds only GS BT phones recognition funcionality.
tftpd-hpa does not correct handle the OPTION parameters in the TFTP packet ;(
At first I tired to implement it into tftpd-hpa, but after
On Thu, 2006-02-09 at 21:27 +0100, Joe wrote:
Greetings to All,
I hope someone has already gotten this working. I spent all day today trying
to get ooh323 and gnugk to run on the same box. After a lot of tweaking to
get everything compiled, I got both up and running.
I've it working. Can
On 2/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Thanks!
Do you have any suggestions on what I might do next. I have the phones, I
have asterisk, and I have everything setup. But i can't get the login to work
with the Polycom function. Nothing happens...and I can't find any readmes'
Hi Rich,
Been using mpg123 for moh for the last two years or so. However, when
I have * config errors, often times get a endless stream of console
messages and need to kill the two mpg123 processes.
Is there an alternative to mpg123 that eliminates that issue?
I see references in
On Fri, 2006-02-10 at 00:45 +0500, Hussain Umair wrote:
hi ive been tryin to get oh323 to work and installed it without any problems
but it gives me the same error all the time this is the third time ive
installed it..please if anyone can kindly help me out thanks in advance...
On Thu, 23 Feb 2006, Ralf Schlatterbeck wrote:
Hello Armin, hello List
I'm trying to get chan_capi working with asterisk from debian stable
(asterisk 1.0.7, the debian version number is 1:1.0.7.dfsg.1-2).
I managed to get it compiled by providing my own version of
ast_copy_string.
Hmm, this
Hi All,
I am a newbie to Asterisk and I was able to install Asterisk and call out.
Recently I purchased two DID's, can someone please tell me or point to some
links showing how to configure these DID's for SIP based softphones like
Express talk?
Thanks,
Manoj.
Hello Dinesh,
sorry for the late reply but that was exactly what was wrong..
using a real 'null' value solved my problem..
thanks,
Bart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dinesh Nair
Sent: zaterdag 4 februari 2006 4:53
To: Asterisk Users
I used madplay with * 1.0 and moved to native for playing mp3's with 1.2 with
no problems. Depends what you want to play, doesn't native stop when there is
no one to play to then restart when there is someone to play to? Might be a
problem if you want to plays ads and don't have many callers,
hi ,
i have some options we are working with at vast deployement with no problems:
www.tigernetcom.com - type 102 is a nice ATA, like GS 486 but far away better.
we import directly from the producer at great prices so if anybody interested, please contact off-list.
additionaly, we know another
Many thanks to everyone for their input. We have been using sipura 1001
and 2002 units and they work great as a SIP adapter, but something that
can also function as a router would be more useful to us. Does anyone
have any comments on the Sipura 2100?
What about a battery backup? Time
Manoj,
just look in AMP to Inbound Routing, fill in the DID, define the softphone as extension X and send the call to extension X
Mickey
On 2/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,I am a newbie to Asterisk and I was able to install Asterisk and call out.Recently I purchased
Hi,
I have problems while trying to dial from simple analog phone that
attached to my TDM400P card.
No matter which number i press i immediately get a congestion tone.
when calling from outside (e.g cellphone )to the line on port 4 and
pressing extension #123 everything works fine and i manage
Can we patch the stable release with your SVN branch
?
Regards
Harry
--- BJ Weschke [EMAIL PROTECTED] a écrit :
On 2/23/06, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Thanks!
Do you have any suggestions on what I might do
next. I have the phones, I have asterisk, and I have
In the most basic case you create a SIP user and create extensions that
point to those SIP users.
in sip.conf:
[sipuser1]
username=sipuser1
secret=123456
type=friend
host=dynamic
disallow=all
allow=ulaw (-- put your most preferred codec here)
allow=gsm (-- other codecs you will support on
On 02/23/06 21:15 Rich Adamson said the following:
I see references in musiconhold.conf relative to madplay, native file
format, asterisk-addons, etc. Not sure why the asterisk-addon approach
hasn't been moved into trunk, or if madplay is a better choice on this
i think it would be better off
When a call is placed out the Zap interface there is a long pause followed
by an error message from the telco that the call can not be placed as
dialed.
We have a tdm2413e with 11 1FB (POTS) lines. The number being dialed is a
working local number, all dialed numbers get the same error.
What
http://cyber-telecom.net/store/product_info.php?cPath=21products_id=34
How about this
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett
Sent: Thursday, February 23, 2006 10:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
I recently moved all of my conf files over to a new Asterisk 1.2.4
server and every works except the features enabled in features.conf. Was
there a syntax chnage in 1.2.4? Or is there something else... Here is my
features.conf:
[general]
parkext = 880;
I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000.
The codec order on each one is the next:
SJPhone: GSM - iLBC - PCMA - PCMU
GXP2000: G729 - GSM - PCMA - PCMU
(I have a G729 license, so there's no problem with transcoding G729)
In my sip.conf, I've defined the following
Hi,
I am getting repeated messages in my logs with the following:
*
Feb 23 07:56:11 NOTICE[2470] pbx.c: Cannot find extension context 'default'
Feb 23 07:56:11 DEBUG[2470] chan_sip.c: SIP message could not be
handled, bad request: [EMAIL PROTECTED]
Feb 23 07:56:12
On Wed, 2006-02-22 at 21:44 +0200, [EMAIL PROTECTED] wrote:
Hi, i'm using an elesign voip gateway esc1700 to call to one iax
sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when
I make the call using the esc1700 the communication is dropped, this is
the log portion:
Dinesh Nair wrote:
On 02/22/06 11:08 C F said the following:
http://bugs.digium.com/view.php?id=5574
That is a patch that will do just that.
while an app is nice, followme could have been done thru some nifty
dialplan work as well.
True, but why not accept the app? It sure makes the
I vote for the raw file format, due to the reasons listed here:
http://www.orderlyq.com/asteriskqueues.html
Also there is a note on the same page as follows:
Note: As of Asterisk 1.2.0, Mpg123 is no longer used by Asterisk, which has
its own MP3 player. However, it's still a good idea to play
You don't need the Polycom ACD support in order to do ACD logins with Polycom
phones. Just dial an extension and call AgentCallBackLogin(). You won't get any
visual confirmation on the phone however of being logged in, but you will be.
If you set the acd-login fields in the phone's xml, the
This
has worked for several months for us. It's /etc/dhcpd.conf
ddns-update-style ad-hoc;
authoritative;option option-66 code 66 = string;
subnet
172.32.16.0 netmask 255.255.255.0
{ #range 192.168.10.101
192.168.10.120; default-lease-time
600; max-lease-time
7200; option
option-66
I had a similar problem, basically asterisk would be sending dialled
digits before the telco was ready to pick them up, in my case some
numbers would work but most wouldn't and i'd get messages from the telco
like you described, the solution I'm using: put a delay before dialing
the number:
from
hi Michael !!
thanks to your support! with the help of a tecnichan of my carrier
i've found the error! the problem was that i have a E1 with 30
channels, but actually only 15 channels are available, 5 only in
inbound (from 1 to 5) and 10 only in outbound (from 22 to 31) so when
i try to call with
When a call is placed out the Zap interface there is a long pause followed
by an error message from the telco that the call can not be placed as
dialed.
We have a tdm2413e with 11 1FB (POTS) lines. The number being dialed is a
working local number, all dialed numbers get the same error.
Johnathan Corgan wrote:
btb wrote:
[7508] ;ipkall
type = peer
dtmfmode = rfc2833
context = remote
callerid = ipkall incoming 7508
nat = no
You've configured this entry as a peer, which is for dialing out, versus
as a user, which is for incoming calls. Solution is to change to
'type=user'.
On 02/23/06 21:15 Rich Adamson said the following:
I see references in musiconhold.conf relative to madplay, native file
format, asterisk-addons, etc. Not sure why the asterisk-addon approach
hasn't been moved into trunk, or if madplay is a better choice on this
i think it would be
Time Warner providesan emta not an ATA and the technology is different. You do not even need internet connection for that and runs over their own private network through DOCSIS. You need to be a cable service provider to afford that. the good ATA that we use is Linksys Rt31P2-NA, make sure you
Yes of cource. But that's not what I'm interested in. I want to be able to see
on the phone if the agent is logged on or not. Automatic logon is not an option
either.
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Douglas Garstang
Skickat:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have actually modified AMP to store the mac address and auto build the
phone.cfg and 0004XXX.cfg files for ftp. I use the default
username and password for the phones, so litterally all you do is plug
them in...
I will put together a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Is it possible to pickup a call that is on hold on another extension?
Does anyone have any magic they can share on this topic?
I am struggling to teach call parking at a local shop where we installed
*. It would simplify my life so much if they
I think I'm very close to getting IAXModem and Hylafax going, but my
current inbound hylafax logs show this:
Feb 23 10:09:37.98: [ 3638]: MODEM Empty line
Feb 23 10:09:37.98: [ 3638]: MODEM TIMEOUT: waiting for v.21 carrier
Two questions -
1) Does anyone know what step I missed here? (I.e.
This is dhcpd.conf which works great with
Polycom 501:
option domain-name blah.com;
option domain-name-servers 192.168.80.3;
default-lease-time 7200;
max-lease-time 14400;
authoritative;
# ad-hoc DNS update scheme - set to
none to disable dynamic DNS updates.
ddns-update-style
I have ISDN2 and 10 MSN numbers and also have just upgraded to 1.2.4 and
chan_capi-cm and have this working completely perfectly
Capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=en
[ISDN1]
isdnmode=msn
incomingmsn=*
controller=1
softdtmf=1
accountcode=
Thanks Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood
Sent: Wednesday, February 22, 2006 6:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Tormenta CAS signaling
Viktor Tatianin wrote:
Hi Damon -
Has anyone been able to get the IP501 to discover the FTP server IP
address (via dhcp or dns) and download 100% of the config from a
provisioning server?
Sure, works great! I'm not sure if you got the TFTP config from the
gentleman who suggested it, but this is really dependent
Hi Again Damon -
I just remembered that the FTP server setup can be tricky, too. The default
username has capitalized letters, and this doesn't work with a lot of FTP
servers. I had to use ProFTPd to get it done. I created a user account
called plcmspip, and added the following to
Yep, been there too.
Most cases are MS IIS, so username case is ignored.
Damon
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 23, 2006 9:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Damon Estep
Subject: FW: auto
On 2/23/06, Darrick Hartman [EMAIL PROTECTED] wrote:
Dinesh Nair wrote: On 02/22/06 11:08 C F said the following:
http://bugs.digium.com/view.php?id=5574 That is a patch that will do just that. while an app is nice, followme could have been done thru some nifty dialplan work as well.True, but
Bob McDowell wrote:
I think I'm very close to getting IAXModem and Hylafax going, but my
current inbound hylafax logs show this:
Feb 23 10:09:37.98: [ 3638]: MODEM Empty line
Feb 23 10:09:37.98: [ 3638]: MODEM TIMEOUT: waiting for v.21 carrier
Two questions -
1) Does anyone know what step I
On 2/23/06, Olle E Johansson [EMAIL PROTECTED] wrote:
Isaac Xiao wrote: We have the same issue happened to all Asterisk versions of 1.2.X (I tried all). In CLI, it shows "-- Incoming call: Got SIP response 500
Internal Server Error back from 192.168.2.104". Once you see this msg, the buddy watch
Hi all,
I want to Monitor a call that is already in progress. How can I achieve
this behavior?
For instance, I have a operator and a client speaking with each other. I
want to listen an record this conversation.
I hope I made myself clear. :)
Thanks in advance.
Fernando Lujan
Hey everyone, I haven't seen an issue quite like mine, so I am hoping
anyone who used the Polycom 601's may have an idea.
We are going to be switching our office over to Asterisk. All the phones
are going to be 601's, I am going to set up a boot server, but for now I
am just going to test
Real Funny, you stated in one of your previous post that you do this ALL the
time
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Wednesday, February 22, 2006 10:06 PM
To: Asterisk Users Mailing List - Non-Commercial
Hi,
I've got the latest [EMAIL PROTECTED] setup running with SIP. I've been
trying to tie it in with the PSTN off and on for a while with no
success. I gave up on ISDN and purchased 2 x100p cards from x100p.com.
I've got the card installed, the machine can see it.
The problems I have at the
Im trying to detect when the person I call pick up the phone and also
when I (and the other side) hang up. Until now I've tried with the G
option in the Dial app. wich one continues running dialplan when the
other side picks up, but I have no option to know when he hungs up.
Other option
I have to provision several dozen * users to a seperate building on our
campus in the same subnet. Ordinarily, I'd just run a gigabit cat6 cable to
another switch if it doesn't violate the 100 metre rule, but this building
is several hundred metres away from my backbone. My only option for cabling
If you can read Spanish, check
http://blog.julianmenendez.es/asterisk-hylafax-iaxmodem
Julian.
On 2/23/06, Bob McDowell [EMAIL PROTECTED] wrote:
I think I'm very close to getting IAXModem and Hylafax going, but my
current inbound hylafax logs show this:
Feb 23 10:09:37.98: [ 3638]: MODEM
Looks like you havn't read the notes on that bugs. Anyhow lots of apps
in asterisk could have been done in the DP, Page comes to mind.
On 2/23/06, Dinesh Nair [EMAIL PROTECTED] wrote:
On 02/22/06 11:08 C F said the following:
http://bugs.digium.com/view.php?id=5574
That is a patch that
In addition, using the dialplan to do this
can make a mess of your CDRs, whereas an application can take better control of
thatsituation.
I'll
second that, I have implemented a followme in my dialplan and my CDR's require
creative SQL queries in order to determine definitively a user's
Bob McDowell wrote:
I think I'm very close to getting IAXModem and Hylafax going, but my
current inbound hylafax logs show this:
Feb 23 10:09:37.98: [ 3638]: MODEM Empty line
Feb 23 10:09:37.98: [ 3638]: MODEM TIMEOUT: waiting for v.21 carrier
Two questions -
1) Does anyone know what step I
For the sake of the list's archives, my problem was rxgain. I used
ztmonitor to tweak the gain until I got at least three '#'s on the RX
side during a fax.
Thanks,
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob
McDowell
Sent:
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