Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias
On 15/11/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias
Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias
On 15/11/06, Mattias Andersson [EMAIL PROTECTED] wrote:
Hi!The problem is that I have commercial Asterisk baste switch that it
Hi,
When a call is made to a queue and picked up by agents at least 2 CDR
entries are made, one from local to the agent's (sip) phone, and from
incoming line to Agent. There are other entries generated when other
conditions happen, like agent do not pickup phones and so on.
Going through the
[EMAIL PROTECTED] said:
I'm trying to set up the Music on Hold feature.
However, when I place a call the moh starts and stops
immediately and as a result I dont hear the audio.
On 2006-11-13 00:14:40 -0800, zen Perry [EMAIL PROTECTED] said:
Mac OS X, Asterisk 1.4 beta
Yeah, I am
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Peter Howard
Sent: 14 November 2006 20:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: Problem found Re: [asterisk-users] Headaches
with Video over SIP
Codec
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
James R. Stevens
Sent: 14 November 2006 20:36
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] In the beginning-The first question.
List,
Im a Cisco certified Network guy with little
Hi Friends,I have installed Asterisk and configured successfully. Now, I got a doubt. Here I am giving my configuration.1) 1 PSTN line connected to FXO port and created inbound route. (Ph. No: 233534)2) 1 Analog phone connected to FXS port and created ZAP extension with No. 1033) Configured
Hello Users,I googled Call Snooping, its shows only the it means, But i didn't find How to dialplan the Call Snooping,I seen that What is Trixbox in Asterisk I Use only some Feature in Asterisk (20),
Is it need Asterisk to install the TrixBox in that same System where i installed the Asterisk
Ciao Matthew,
What do I have to do, exactly, to install Meetme?
You have to build Zaptel before building Asterisk, because MeetMe uses
Zaptel modules for timing. Then, when you build Asterisk the MeetMe app
will automatically be built.
See http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias
Sorry if I have made double post! (Difficult to verify if mail was sent).On 15/11/06, Darryl Dunkin [EMAIL PROTECTED]
(I'm in the UK if that makes a difference)
There seems to be a plethora of different ISDN cards available in both the
BRI and PRI range - all with varying prices too - from £25 to nearly £1000
from some popular reseller sites...
Does anyone have (or know of) a good comparison site, or have
chanspy
see: http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy
-VijOn 11/15/06, raviprakash sunkara [EMAIL PROTECTED] wrote:
Hello Users,I googled Call Snooping, its shows only the it means, But i didn't find How to dialplan the Call Snooping,
I seen that What is Trixbox in Asterisk I Use
Would anyone like to recommend a good and reasonable quality ISDN
card for use in the UK, as after a lot of good results with TDM400P
cards with several systems installed now, I need to look at a few
ISDN BRI (old business highway about to move to ISDN2) and possibly a
single-line PRI
Hi,
I want to change my voicemail message based on the time of day. I would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.
I have come up with two ways of doing it:
1. A cron job to replace the files
Hi,
I want to change my voicemail message based on the time of day. I would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.
I have come up with two ways of doing it:
1. A cron job to replace the files
Just use two different contexts for the two times of day (open/closed)
and use Playback to play the correct message before going direct into
voicemail without any prompt.
M
Wildheart wrote:
Hi,
I want to change my voicemail message based on the time of day. I would
like a message that
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
Hi all,I want to initiate a call from the asterisk to an extension, where I will forward the asterisk side to another extension later (to the conference extension). I can initiate a call uning originate call from an extension to the desired extension, but it would need someone from the
On Wed, 2006-11-15 at 11:23 +, Senad Jordanovic wrote:
Would anyone like to recommend a good and reasonable quality ISDN
card for use in the UK, as after a lot of good results with TDM400P
cards with several systems installed now, I need to look at a few
ISDN BRI (old business highway
Isn't that covered in point 2? Admittedly, I did not consider using
Playback rather than voicemail to play the message. But you didn't point
that out anyway.
a lil bit of googling wud have answered you Tim.
Put in some effort next time anyway, for now :
In article [EMAIL PROTECTED],
Sharon Lim [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
-=-=-=-=-=-
Thanks, will do more research on that part. By the way, Im trying to do IVR
where caller enter the pin the retrieve some information out of the MS SQL.
I am wondering, what is the constraints or how to
Beronet cards have 2 or 4 ports are very good.Those guys produced the misdn driver, that is now Digium uses for their new BRI card.www.beronet.comtheir tech support has been very very good.
On 11/15/06, Conrad Wood [EMAIL PROTECTED] wrote:
On Wed, 2006-11-15 at 11:23 +, Senad Jordanovic wrote:
On Wed, Nov 15, 2006 at 11:15:16AM +, Gordon Henderson wrote:
(I'm in the UK if that makes a difference)
There seems to be a plethora of different ISDN cards available in both the
BRI and PRI range - all with varying prices too - from ?25 to nearly ?1000
from some popular reseller
Hi
I am testing Asterisk connecting to our Alcatel 4400 PBX. I have a
wcte11xp card - all is well - but we cannot communicate with the Alcatel
- when we try to call an Alcatel extension, we see Error 34 no channels
available on the CLI.
I suspect that this is because of invalid span and
I have problem with fax machine Panasonic DX600. It's connected to Grandstream
Handy Tone 386 which is connected to Asterisk. Asterisk is connected to my SIP
provider.
To some numbers I can't send FAX, and I get following error on CLI.
WARNING[2237] chan_sip.c: Unknown SDP media type in offer:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I am seeing the following in my log file (standard trixbox install).
One seems to be complaining about an error in the dialplan but it
won't tell me what file or what line. The other (maybe related) is
complaining about a channel lock.
Hi all.
I have enabled the attended transfer feature in features.conf. I'm
using it and I want to resolve some questions, I hope someone can help
me :)
When I transfer a call to an extension:
- The extension rings during 15 seconds and the call returns to the
transferer. Is there any
I have an IVR for employees to enter certain information, like employee number and such and then I pass that to a simple agi/php script that build the query string and uses freetds. It took me a while to get it working and reproduce it on several systems, but I am rather new to Linux in general.
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
(a) If you are not running a version of Asterisk that has working SIP
jitter buffering (is there such a thing?), then abandon all hope now.
(b) We have no experience with the Cisco ATAs, but the Linksys (nee
Sipura) SPA-210x is
Hi,It seems AVM C2 and C4 ISDN-BRI active boards are not distributed anymore (is true everywhere).Eicon-Dialogic boards seem to have good Asterisk support, thanks to chan-capi.What are the best other CAPI-compliant boards (with embeded fax DSP) one could use with Asterisk ?
Regards
On 10/10/06, LJ [EMAIL PROTECTED] wrote:
In my Asterisk 1.2.9.1 installation I use the following:in voicemail.conf
include the following:exitcontext=vmloginoperator=yes
Sorry to revive a month old thread but here was the easy button solution for me.
With debugging on I did a reload
Hello
I want build big asterisk server. Server will be work as gateway between PSTN
and VoIP network. I think about 1000 SIP users accounts and 4 E1 ports. I know
that preformance in this case depend on codeck which will be use. I want use
card with CAPI interface. Can you describe me your
Is there a way to make Asterisk don't send 482 Loop Detected error
messages and continue with the transaction that is taking place?
Thanks,
Ricardo.
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asterisk-users mailing list
To
Ricardo Carvalho wrote:
Is there a way to make Asterisk don't send 482 Loop Detected error
messages and continue with the transaction that is taking place?
Not that I know of since a loop is an error.
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Func_odbc (which is new in 1.4) was
backported to 1.2. See http://www.asterisk.org/func_odbc
While it only will return one row (there
are patches to make it return multiple rows), its very useful for our
purposes. You set up the function in func_odbc.conf, call it with
For the Asterisk side of things, are you using Asterisk directly or Trixbox? I'm just trying to get a prototype working so don't want to spend a lot of time on the initial asterisk setup. If Trixbox will allow me to do the php+agi integration, I'll do that, if not, will try to just try to install
Title: SIP NOTIFY routing problem
In version 1.2.7.1 I have an endpoint (number 5302) registered.
'sip show peer 5302' shows that the Reg. Contact address is:
sip:[EMAIL PROTECTED]:5066
When I call 5302 I see INVITE messages correctly routed to the contact address with request lines
On 11/15/06, Wildheart [EMAIL PROTECTED] wrote:
Hi,
I want to change my voicemail message based on the time of day. I would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.
I have come up with two ways of
Hi
guys,
i would check the
state of a number on a Zap channel, i suppose that i cannot use ExtensionState
that works only for SIP and IAX.
Anyone has any ides
? Could i check the state of a pubblic number before transfer it a internal
call?
Thanks in
advance
Giordano
--
No virus
What you *could* do is record one greeting as the unavailable message and another as the busy message and during the day, just play the unavailable one and at night play the busy one...
On 11/15/06, C F [EMAIL PROTECTED] wrote:
On 11/15/06, Wildheart [EMAIL PROTECTED] wrote: Hi,I want to change my
Tom Lynn wrote:
Ron,
The guy is trying to help you. Go to the link and read it. There is
a feature that you can use to play a recording into the voice
channel. Mine is set so when you press #9, the caller hears the lots
of monkeys recording.
The best part of it is that you can hang up
I current have a working Asterisk 1.2.12 server with ODBC voicemail storage,
realtime static maps for voicemail, sip and iax configuration files.
Realtime extensions, etc. All works great. I have verified that this
configuration works on my test server as well. Now I am trying to test the
1.4B3
Thanks for the reply. There's
no /usr/lib/asterisk/modules/app_meetme.so , though that dir has all the
libraries for all the other modules I see in CLI 'show modules' (no
meetme there, either, as I noted). /etc/asterisk/modules.conf starts
with
[modules]
autoload=yes
and there's no
yes, you can use Trixbox.On 11/15/06, blackwater dev [EMAIL PROTECTED] wrote:
For the Asterisk side of things, are you using Asterisk directly or Trixbox? I'm just trying to get a prototype working so don't want to spend a lot of time on the initial asterisk setup. If Trixbox will allow me to do
There's no other way? Surely I can't be the first person that hasn't
wanted to do this before.
On 11/14/06, Justin Newman [EMAIL PROTECTED] wrote:
You need to modify app_queue.c to hold off on bridging until the receiving
party has accepted the call. If the receiving party rejects (hangup,
Hi,
I have some trouble with setting my CallerID if i make an international
Call. No Problems with National Calls, i can set whatever I want. We pay
for this service but our telephone provider was not able to state clear,
wether the number we set on an international call should be shown on the
There is a company that I call that requires a * be dialed to break out of
their IVR.
The problem is Asterisk is grabbing that * for itself. Is there a way to get
this sent?
asterisk1*CLI show features
Builtin Feature Default Current
--- --- ---
Hi!
I have configured the language support in asterisk to reproduce spanish
prompts. I have lines for it in sip.conf, iax.conf, zapata.conf and
voicemail.conf as shown:
[general]
...
language=es
...
In zaptel.conf
loadzone = es
defaultzone = es
When I check my voicemail I get in the CLI:
Edwin wrote:
I current have a working Asterisk 1.2.12 server with ODBC
voicemail storage, realtime static maps for voicemail, sip
and iax configuration files. Realtime extensions, etc.
All works great. I have verified that this configuration
works on my test server as well. Now I am
On Wed, 2006-11-15 at 13:54 +0100, Antonio Almodóvar wrote:
Hi all.
I have enabled the attended transfer feature in features.conf. I'm
using it and I want to resolve some questions, I hope someone can help
me :)
When I transfer a call to an extension:
- The extension rings during 15
Matthew J. Roth wrote:
Is it possible to record outbound analog calls using an X100P?
I was asked if I knew how to record all calls for a shop with 4 analog
phones transparently to the end users. I thought Asterisk was a good
fit for this and I envisioned using either Digium TDM400Ps or
I'm trying to use a
simple page function. It starts a MeetMe conference with the devices I've
listed, but the devices hang up after 3-5 seconds. After doing some
research I found this was a problem, and I needed to remove a (5) from
app_page.c
Well, my app_page.c
didn't have the (5). I
On Wed, 2006-11-15 at 17:50 +0100, Tobias Wolf wrote:
Hi,
I have some trouble with setting my CallerID if i make an international
Call. No Problems with National Calls, i can set whatever I want. We pay
for this service but our telephone provider was not able to state clear,
wether the
As per ManxPower at #asterisk, it is not possible to record a call
dialed from an analog phone connected to the Phone In port of an X100P
because the two ports on the card are hard-wired together.
A bit off-topic maybe, but does that then mean you can't
make 2 simultaneous calls through
On Wed, 2006-11-15 at 14:41 +0100, doki_cti wrote:
Hello
I want build big asterisk server. Server will be work as gateway between PSTN
and VoIP network. I think about 1000 SIP users accounts and 4 E1 ports. I
know that preformance in this case depend on codeck which will be use. I want
use
Title: dtmf tones not always recognized
We have analog phones (Aastra 390) connected to channels banks (Adtran TA750) connected to a 4 port digium card( TE410P).
Because of echo problems we purchased external T1 echo cancellers from Orion Telecom. (The TE412P did not eliminate enough of
BAH!
My Makefile in the apps folder was missing
app_page.c. I added it, recompiled, page is working
properly.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
WilliamsSent: Wednesday, November 15, 2006 10:33 AMTo:
Asterisk Users Mailing List - Non-Commercial
chan_sip.c: Got 200 OK on REGISTER that isn't a register.
i'm getting the above warning
while trying to register a phone from outside of asterisk network.
( so no registration what so ever, no dial tone and what not)
it registered once for about 20 minutes
exepted calls and i could call out
but
Hello,
I have an Asterisk system with kernel 2.6.18.1 and one quadbri. I
have installed the latest bristuff patches (0.3.0-PRE-1s). The system works
fine, but when I do a reboot, the system hangs unloading module qozap.
Is there any known problem with latest 2.6 kernels and qozap
Ken Williams wrote:
I'm trying to use a simple page function. It starts a MeetMe
conference with the devices I've listed, but the devices hang up after
3-5 seconds. After doing some research I found this was a problem,
and I needed to remove a (5) from app_page.c
Well, my app_page.c didn't
We use intercom 100%
inter-office. To get FreePBX to do this with Grandstreams by default
without having to create intercom or paging groups, just change the following
line (line #58) in your extensions.conf from:
exten = s,10,Dial(${ds})
; dialparties will set the priority to 10 if $ds is
as far as i know sip-based
i came across something that said
this could be due to too much traffic
but the mesage was not clear on what side
Original Message
Subject: Re:[asterisk-users] Got 200 OK on REGISTER that isn't a register
From: Ron McLeod [EMAIL PROTECTED]
To:
I think this message is saying that it received a 200 OK for a REGISTER
message that Asterisk does not know about (anymore).
Is you system trying to register with an ITSP or other SIP-based system?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
i figured that out
what i can't find is a solution to the problem
Original Message
Subject: Re:[Asterisk-Users] Got 200 OK on REGISTER that isn't a register
From: BJ Weschke [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
We have 1 server that after a few hours operating has multiple process
of asterisk running. Here is the pstree output:
# pstree
init-+-atftpd
|-auditd---{auditd}
|-bash---safe_opserver---op_server.pl
|-crond
|-cwASTcall.pl
|-dbus-daemon
|-events/0
I am trying to get the example input.php working from PHPAGI but it will not playback the letters that I put in because of this error:Nov 15 14:25:22 WARNING[18678] file.c: File /tmp/swift_f87b365372c500c76e497087ac7e470a does not exist in any format
But the file does exist and I see the entries
On Thursday 16 November 2006 06:44, Conrad Wood wrote:
On Thursday 16 November 2006 06:42, Matthew J. Roth wrote:
As per ManxPower at #asterisk, it is not possible to record a call
dialed from an analog phone connected to the Phone In port of an X100P
because the two ports on the card are
Are you including the file extension?
Jay
Tom Vile wrote:
I am trying to get the example input.php working from PHPAGI but it will
not
playback the letters that I put in because of this error:
Nov 15 14:25:22 WARNING[18678] file.c: File
/tmp/swift_f87b365372c500c76e497087ac7e470a does not
There are no file extensions. It is just-rw-r--r-- 1 asterisk asterisk 32 Nov 15 12:52 swift_082da06a422be49e3a475925d9fc50e6-rw-r--r-- 1 asterisk asterisk 7 Nov 15 12:52 swift_6fc422233a40a75a1f028e11c3cd1140
-rw-r--r-- 1 asterisk asterisk 13 Nov 15 12:52
Hi Gustavo -
I just received two TDM400P cards, but I'm having problems with them.
On a x86 stable Gentoo box.
Kernel: 2.6.17
gcc-4.1.1, glibc-2.4-r4
Is that an hardware problem? Should I try the other card?
I tried the other card and the problem is still there. REALY NEED HELP
Has anyone noticed that attempting to place a call from the Placed
Calls list on a Polycom IP501 by pressing the 'Dial' softkey
sometimes
simply returns the phone to the idle screen?
Yes, I've seen it. We're running 1.6.6, what firmware version do you
have?
We're running SIP 1.6.6.0036
its normal .if there are many calls going . You should worry if your load or memory usage is very high .On 16/11/06, Andre Courchesne - Consultant
[EMAIL PROTECTED] wrote:
We have 1 server that after a few hours operating has multiple processof asterisk running. Here is the pstree output:#
Hi all,
I have installed this package onto my Debian and placed the files i want the
Cisco 7960 phone to get from the tftpdboot directory. But it doesn't seem to
work. Are ther any special settings I should do to this server?
Many thanks for all your help,
Christian
1) Try giving it an extension (say .gsm) and seeing if that works. Make
sure you change both the file and your script.
2) Does the rest of the script work? If you run './test.php', do you
get any errors?
Jay
Tom Vile wrote:
There are no file extensions. It is just
-rw-r--r-- 1
Is it possible to record outbound analog calls using an X100P?
I was asked if I knew how to record all calls for a shop with 4 analog
phones transparently to the end users. I thought Asterisk was a good
fit for this and I envisioned using either Digium TDM400Ps or Sangoma
A200s with 4 FXO and 4
You may need to create (or modify) the tftp file in /etc/xinetd.d. For
example:
service tftp
{
disabled= no
socket_type = dgram
protocol= udp
wait= yes
user= root
Does anyone have any experience using the Huawei Videophones in a
point-to-multipoint configuration using Asterisk?
Thanks,
Nabeel
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To UNSUBSCRIBE or update options
It does error when I run it from the CLI with error:
sh: -p: command not found
I am assuming that it is referring to this line in phpagi.php
shell_exec({$this-config['cepstral']['swift']} -p
audio/channels=1,audio/sampling-rate=$frequency $voice -o $fname.wav -f
$fname.txt);
On 11/15/06, Jay
I have just installed Asterisk and installed the sample configuration
files. Asterisks appears to be working and I have added a SIP client:
[John]
type=friend
secret=test
host=dynamic
allow=all
I have been trying to dial the demo number 500 when using PortSip,
Asterisks answers the
For the Asterisk side of things, are you using Asterisk directly or Trixbox?
I'm just trying to get a prototype working so don't want to spend a lot of
time on the initial asterisk setup. If Trixbox will allow me to do the
php+agi integration, I'll do that, if not, will try to just try to
Yes, I get same error message in my log. Anybody has any info on this one?
Are you using trixbox? It would be nice to try and isolate this
problem by ruling out a bad config in trixbox.
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I've installed Grandstream GPX-2000 phones and have successfully enabled
one of my buttons to use Asterisk BLF for an extension. I can tell when
this extension is available, is being rung, or is on the line.
I'd like to do the same for my Zaptel channels, to be able to see when a
line is
try :
[John]
type=friend
secret=test
host=dynamic
disallow=all
allow =gsmilbculawalaw
Also try other sip phone slike sjphone just to make sure there is no prob .
On 16/11/06, Charlie Grosvenor [EMAIL PROTECTED] wrote:
I have just installed Asterisk and installed the sample configuration
files.
I've installed Grandstream GPX-2000 phones and have successfully enabled
one of my buttons to use Asterisk BLF for an extension. I can tell when
this extension is available, is being rung, or is on the line.
I'd like to do the same for my Zaptel channels, to be able to see when a
line is
Charlie Grosvenor wrote:
[John]
type=friend
secret=test
host=dynamic
allow=all
Try:
disallow=all
allow=alaw
allow=ulaw
allow=gsm
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
Trixbox scatters it's config files. Some stuff is kept in the
database, some in the conf files.
You have to keep your configuration in specific files that won't be
overrritten.
True - TB does a lot of very specific stuff. If you want to have a
plain Jane dial plan for your stuff then use the
Hi,
I'm sending the following message from port X to port 5060 of another box
running Asterisk, and it is answering back to port X from port 5060. Shouldn't
Asterisk use the Via header to find out where to answer, and in this case send
its answer to port 4000?
OPTIONS sip:192.168.0.103
I am looking for a web interface to control my zap agents. Allowing them
to do conferences and transfers. I am familiar with flash operator panel
but am unsure of how I would set it up to allow the agent, caller, to
dial another number and have a three way conference. I have setup
features.conf to
Upon further investigation I must be doing something wrong.
It was my understanding that a hint extension could be anything, it
wasn't the same as a real extension, though you could make it the same
to make it easier.
That being said exten = 702,hint,SIP/702 works, while exten =
Have you tried setting the CALLERID variables? If the provider is
ignoring those then I guess they are asking you to set per call
blocking? I don't know how to do that.
exten = s,1,Set(CALLERID(number)=3025551212|a)
exten = s,n,Set(CALLERID(name)=Joe Smith|a)
MARK.
Emil Thelin wrote:
Hi!
nat=yes might cause this, since with NAT we cannot trust the IP or the
port that is in the data part of the packet.
Andre Kirchner wrote:
I'm sending the following message from port X to port 5060 of another box
running Asterisk, and it is answering back to port X from port 5060. Shouldn't
On Wed, 2006-11-15 at 13:34 -0700, Andre
Courchesne[EMAIL PROTECTED] wrote:
I remember seeing a ton of asterisk lines in ps if you had just the
exact right (wrong?)
declare in safe_asterisk. I had it myself and erased the line a while
back. I can't see it in the svn repository at all. It was
Hi!
The problem is that I have commercial Asterisk baste switch that it works wit.
My trixbox do not. I guess it has to do with the
setting of system for Caller ID.
//Mattias
I do not now way, but my posting are not coming trow. Or are the?
//Mattias
On 15/11/06, Mattias Andersson
Which version are you using? There was a problem in 1.2.12.1 with the page
application. Update to 1.2.13.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steven
Ringwald
Envoyé : 15 novembre 2006 13:45
À : Asterisk Users Mailing List -
I am new to the Asterisk code, but it looks it the response to OPTIONS is
always sent to the IP address and UDP port that the request was received
from. Also, it looks like Asterisk doesn't deal well the VIA header anyway.
In chan_sip.c it looks like to gives-up if the VIA contains SIP/2.0/UDP:
Thanks, Noah - we'll try 1.6.7 and see if the problem goes away.
CP
On 15-Nov-06, at 11:55 AM, Noah Miller wrote:
Has anyone noticed that attempting to place a call from the
Placed
Calls list on a Polycom IP501 by pressing the 'Dial' softkey
sometimes
simply returns the phone to
On Mon, Nov 13, 2006 at 09:50:08PM -0500, Zeeshan Zakaria wrote:
On checking tail -f /var/log/atftpd, I can see that on reboots, other phones
get served by the TFTP, but not the linksys ones. Now I don't understand how
was it was updating itself and was being provisioned resyncing for so many
Whoops, sorry - it only handles SIP/2.0/UDP; which is what is expected,
but it seems like it only checks for the VIA header for REGISTER, INVITE,
CANCEL, BYE, and SUBSCRIBE requests.
Ron
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre Kirchner
Sent:
Hi,
My case is a little bit complicated.
I would like to use my Asterisk Box for 2 different services/providers :
- Voicemail server for one
- SIP Registrar and Proxy for some other extensions
The problem is that Voicemail service is for another provider which has
defined Extension like ABC ...
A day of banging my head against a wall and spamming this list is about
done
I've got everything working beautifully, and I'm ready to go full out
and implement all across the board...save for one stupid little thing.
We have 6 phone lines and I'd like the GXP-2000 to show the status of
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