Adding "fromuser" option in trunk declaration in AST1 has solved all
problems though.
On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky <[EMAIL PROTECTED]> wrote:
> Rizwan Hisham wrote:
> > I am having a strange problem. I am using my asterisk server AST1 to
> > register with another asterisk
Dear all, I have small question
in sip.conf I added
[service]
type=friend
;username=
;secret=
qualify=900
host=X.X.X.X
dtmfmode = rfc2833
disallow=all
;allow=g729
allow=gsm
allow=alaw
allow=ulaw
and I can proccess incoming call from soft phone only I calling on
number that is used in extensions
On Wed, Mar 5, 2008 at 10:12 AM, Daniel Suleyman <[EMAIL PROTECTED]> wrote:
> but when I use next construction(As I understand it is used to allow
> to process any extension dialed by user)
>
> exten => s,1,Answer;
> exten => s,2,Playback(hello-world,skip);
> exten => s,3,Hangup;
AFAIK, "s"
On Wed, Mar 05, 2008 at 02:12:47PM +0400, Daniel Suleyman wrote:
> Dear all, I have small question
>
> in sip.conf I added
>
> [service]
> type=friend
> ;username=
> ;secret=
> qualify=900
> host=X.X.X.X
> dtmfmode = rfc2833
> disallow=all
> ;allow=g729
> allow=gsm
> allow=alaw
> allow=ulaw
>
>
The idea is that the person connecting and dial anything he want and
the script is deciding to proceed the call or to terminate it(I think
it will be easy to manage extensions.conf - no need to create
extensions).
You know {i} doesent work
exten => 1,1,Answer;
exten => 1,2,Playback(hello-world,sk
Did interupt sharing caused this problem or are you still having this
problem?
After i checked and solved the IRQ sharing I still have this problem.
I use an TDM410B.
I use misdn
Volume settings are default
Matthew Yingling schreef:
> I recently moved an installed and working Asterisk syste
On Wed, Mar 05, 2008 at 04:04:00PM +0400, Daniel Suleyman wrote:
> in curent config i didnt set context in sip(unsing default) but it
> doesent matter if I set up context the same thing s and i doesn't work
>
> now dial plan show next
>
> Context 'default' created by 'pbx_config' ]
> '7007' =
On Wed, Mar 5, 2008 at 12:04 PM, Daniel Suleyman <[EMAIL PROTECTED]> wrote:
> The idea is that the person connecting and dial anything he want and
> the script is deciding to proceed the call or to terminate it(I think
> it will be easy to manage extensions.conf - no need to create
> extensions)
Hi list,
After successfully configuring Linksys SPA3000 and SPA3102 devices as
Asterisk PSTN gateways, the only thing I can't get working is the PSTN
Caller ID. The analog and SIP phones I've used can both display CIDs
for internal calls, while the analog model also displays CIDs
correctly
Hi.
I am trying to pass a variable from one Asterisk PBX
to another.
I'm using DUNDi with IAX2. Is there a way to do it?
I tried the following but it fails.
On peer1:
[dundi-outgoing]
switch => DUNDI/priv
exten => s,1,Set(CDR(userfield)=test)
exten => s,2,Set(DUNDIVAR=${ARG1}#TEST)
exten => s
On Wed, Mar 05, 2008 at 12:41:23PM +, Andres Jimenez wrote:
> On Wed, Mar 5, 2008 at 12:04 PM, Daniel Suleyman <[EMAIL PROTECTED]> wrote:
> > The idea is that the person connecting and dial anything he want and
> > the script is deciding to proceed the call or to terminate it(I think
> > it w
Quoting Jaap Winius <[EMAIL PROTECTED]>:
> Hi list,
>
> After successfully configuring Linksys SPA3000 and SPA3102 devices as
> Asterisk PSTN gateways, the only thing I can't get working is the PSTN
> Caller ID. The analog and SIP phones I've used can both display CIDs
> for internal calls, while
On Tue, Mar 4, 2008 at 7:18 PM, marek cervenka <[EMAIL PROTECTED]> wrote:
> can you recommend "clean&simple&stable" solution for incoming call popup
> (in browser)?
ADM http://adm.hamnett.org/ can invoke browsers when a call arrives.
raj
___
-- Bandw
Hello,
I've found that my Polycom-501 accepts INVITES from any server in the
world... I would like to restrict it to accept calls only from the servers
listed in its config file, but I cannot find anything in the documentation. Any
idea?
Thanks, __Yehavi:
On Wed, Mar 5, 2008 at 1:36 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> This is not needed. If the extension is not found, there is a
> fallthrough to 's' (Right? Or is it chan_zap-specific)?
I would say it's chan_zap-specific.
From
http://www.voip-info.org/wiki/index.php?page=Asterisk+con
HI
I am running asterisk 1.4.18 on both machines.
I have two asterisk machines setup and a polycom phone, the main machine
is #1 the polycom
phone is connected to it as the server.
I come offhook on the polycom, dial 21, which calls machine #2.
Machine #2 is only to play demo-congrats.
Watching t
--- Vieri <[EMAIL PROTECTED]> wrote:
> On peer2:
>
> [dundi-incoming]
> exten => _X.,1,NoOp(Received EXTEN ${EXTEN}.)
> exten => _X.,n,Set(EXTTODIAL=${CUT(EXTEN|#|1)})
> exten => _X.,n,Set(DUNDIVAR=${CUT(EXTEN|#|2)})
> exten => _X.,1,NoOp(Extracted extension ${EXTTODIAL}
> and DUNDi variable ${D
Vieri wrote:
> Hi.
>
> I am trying to pass a variable from one Asterisk PBX
> to another.
>
> I'm using DUNDi with IAX2. Is there a way to do it?
>
> I tried the following but it fails.
>
> On peer1:
>
> [dundi-outgoing]
> switch => DUNDI/priv
> exten => s,1,Set(CDR(userfield)=test)
> exten => s,2
On Wed, 2008-03-05 at 13:20 +0100, Fons van der Beek wrote:
> After i checked and solved the IRQ sharing I still have this problem.
If you're still experiencing problems, you should contact Digium
support. They'd be happy to help you help identify and eliminate the
problem.
--
Jared Smith
Commu
Hi people,
I have a few SPA-942 around, all of them work fine except one. The one
behind NAT..
In every phone you can:
* Pickup a Call on one of the line buttons,
* Create a new call on another button
* Press "xferLx" to join those to calls.
This works everywhere except on
Looks like some nice new features in fonality with their find me/follow
me functionality
http://www.voip-news.com/feature/fonality-find-follow-feature-030408/
love the 'boss overide' call routing - way overdue. I haven't seen the
UI on this yet so cant comment on how well they've implemented i
Hello,
We're attempting to use Asterisk for distributing calls via SIP in a
large-scale speech recognition/VXML environment. We currently use DNS SRV
with weights and priorities to instruct VoIP gateways (not Asterisk) to
route calls to pools of servers. This works extremely well and provides fo
ASTassistant can do this as well. www.astassistant.com
-Scott
- Original Message -
From: "Rajkumar S" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 05, 2008 5:48 AM
Subject: Re: [asterisk-users] incoming call popup
> On Tue,
Quoting Tim Johnson <[EMAIL PROTECTED]>:
> Your caller ID is probably being over-ridden by the settings in your
> sip.conf file. Remove the caller ID from your PSTN section of the
> sip.conf, and the CID should be passed on from the POTS line.
That sounds like a good idea regardless. On the SPA30
Quoting Jaap Winius <[EMAIL PROTECTED]>:
> Quoting Tim Johnson <[EMAIL PROTECTED]>:
>
>> Your caller ID is probably being over-ridden by the settings in your
>> sip.conf file. Remove the caller ID from your PSTN section of the
>> sip.conf, and the CID should be passed on from the POTS line.
>
> Th
Is the new NIN Ghosts music (free download) safe for MOH?
Justin
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now.
http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tA
This is what I have working
topsoftkey5 type: speeddial
topsoftkey5 label: Park
topsoftkey5 value: "##70"
topsoftkey5 states: connected
-Original Message-
From: Russell Brown [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 04, 2008 1:22 AM
To: asterisk-users@lists.digium.com
Subject: Re:
I would advise against it right now and contact the artist. The RIAA
or the (forgot the name, they charge restaurants and stores for
playing music over the PA) would assume that if it is popular one of
their members owns it.
Other than that I would also assume your primary audience must share
you
Ian,
I'm unable to transfer calls using *2, I'm not sure why. Here's my configs:
*sip.conf*
[User1]
type=friend
username=111
context=default
callerid=User Name <111>
host=10.10.1.111
nat=no
canreinvite=no
dtmfmode=info
call-limit=4
[EMAIL PROTECTED]
disallow=all
allow=ulaw
allow=alaw
;allow=gsm
When attempting to build zaptel I get the following error:
configure:2184: error: C compiler cannot create executables
vi config.log
configure:2066: $? = 0
configure:2073: gcc -v >&5
Using built-in specs.
Target: i386-redhat-linux
Configured with: ../configure --prefix=/usr --mandir=/usr/share/man
--- Richard Lyman <[EMAIL PROTECTED]> wrote:
> Vieri wrote:
> > Hi.
> >
> > I am trying to pass a variable from one Asterisk
> PBX
> > to another.
> >
> > I'm using DUNDi with IAX2. Is there a way to do
> it?
> >
> > I tried the following but it fails.
> >
> > On peer1:
> >
> > [dundi-outgoing]
Hi list,
I'm wondering if it's possible to transfer a call that is still ringing??? I
Have some Grandstream GXP-2000 and with the TRNF button it's impossible. So,
I've configured some keys to transfer the calls like this:
[featuremap]
blindxfer => #2; Blind transfer (default is #)
di
That'd be ASCAP (I think there's another one too). They're the ones known for
calling up places, asking to be put on hold to listen to the hold music, then
querying on whether it's been licensed or not (among other tactics).
Pretty much, unless it's music developed in-house, I wouldn't put it
Hi,
I'm currently trying to connect an Aastra 57i to our Asterisk Server.
The strange thing is, that altough I have definitely entered the correct
IP address of the server, the phone doesn't even attempt to register.
Here is the configuration file (local.cfg) of the phone:
firmware md5: dee6e93
Greetings,
The software G.729 codec module from Digium has been updated for all platforms.
There are x86_32 and x86_64 versions optimized for specific processors
available for both Asterisk 1.6 and 1.4 for the following platforms.
* Linux
* Solaris 10
* FreeBSD 7.0
* FreeBSD 6.1
Changes
CSB wrote:
>
>When attempting to build zaptel I get the following error:
>configure:2184: error: C compiler cannot create executables
>
on a Debian/Ubuntu-system
apt-get install build-essential make bin86
solved this problem for me.
Stefan
--
in-put
On Wed, Mar 05, 2008 at 09:10:30AM -0800, Justin Newman wrote:
> Is the new NIN Ghosts music (free download) safe for MOH?
Did you read the license?
http://ghosts.nin.com/main/faq
Ghosts I-IV is licensed under a Creative Commons Attribution
Non-Commercial Share Alike license.
http://creativ
Hi All;
I need a help as the voicemail need GSM codec while I
am using G729 for the call, why Asterisk does not do
codec translation from G729 to GSM, it does not
support?
Any need for settings, what I am missing?
Regards
Bilal
___
You need to configure the proxy address to be the same as the registrar
address.
Gareth
---
This email communication may contain CONFIDENTIAL, PRIVILEGED and/or
LEGALLY PROTECTED information and is intended only for the name
At 09:30 AM 3/5/2008, you wrote:
>This is what I have working
>
>topsoftkey5 type: speeddial
>topsoftkey5 label: Park
>topsoftkey5 value: "##70"
>topsoftkey5 states: connected
On my 480i CT I'm using:
softkey11 type: park
softkey11 label: "Park Call"
softkey11 value: asterisk;700
softkey11 states
On Wednesday 05 March 2008 12:35:53 bilal ghayyad wrote:
> I need a help as the voicemail need GSM codec while I
> am using G729 for the call, why Asterisk does not do
> codec translation from G729 to GSM, it does not
> support?
>
> Any need for settings, what I am missing?
If you do not have a li
Hi,
The Solaris build still appears to be at v32. Am I being a little hasty :)
Thanks
Bruce
The Asterisk Development Team wrote:
> Greetings,
>
> The software G.729 codec module from Digium has been updated for all
> platforms.
> There are x86_32 and x86_64 versions optimized for specific pro
Quoting Tim Johnson <[EMAIL PROTECTED]>:
> What do you have for your "PSTN Answer Delay" (in PSTN tab)? I had to
> set mine between 3 to 5 to get reliable CID from the POTS line. This
> was for a SPA3102, not a 3000. I've never had a 3000, but everyone
> says they are nearly identical.
I normally
Hi All;
I have the following configuration in my iax.conf
files at asterisk box1 and box2 (two asterisk):
At box1:
[user1]
disallow=all
codec=g729
codec=GSM
At box2:
[user2]
disallow=all
codec=g729
codec=GSM
If G729 is no more available at box1, so how can I let
user1 to select GSM codec inst
Stefan,
There are a couple of things you'll want to look at but I would
seriously recommend setting it back to factory default and starting
over. This ensures the settings I don't mention below (a couple you have
changed) are reset to factory.
If there is a specific reason to have different exten
On 06:50, Thu 06 Mar 08, CSB wrote:
> When attempting to build zaptel I get the following error:
> configure:2184: error: C compiler cannot create executables
>
> I believe I have all the necessary packages installed.
>
> Having done some research, one link suggests using strace and in that cas
On Wed, Mar 05, 2008 at 07:26:21PM +0100, Stefan Guenther wrote:
> CSB wrote:
> >
> >When attempting to build zaptel I get the following error:
> >configure:2184: error: C compiler cannot create executables
> >
> on a Debian/Ubuntu-system
>
> apt-get install build-essential make bin86
$ apt-cache
On Thu, Mar 06, 2008 at 06:50:40AM +1300, CSB wrote:
> I believe I have all the necessary packages installed.
>
> Having done some research, one link suggests using strace and in that case I
> don't get the error:
> strace -f -o /tmp/trace -e trace=process ./configure
> ...
> configure: *** Zapte
I setup a number of remote phones on public IPs using the web
interface. Now my question is how do I change the default Polycom:456
password via the web interface. Is there a hidden way or does it have
to be done via FTP TFTP?
Thanks,
Steve Totaro
___
To solve the problem of Local channels answering Queue calls, I thought
about myabe using a channel variable switch that turns on before the Queue
is called, and a check in the extension-dial to Local/ext to see if it is a
queue call and shouldn't go to voicemail, or if it's just a direct call that
I have an unusual and recurring problem since I upgraded to Asterisk 1.4.
Sometimes, mid-way through a call, I hear 6 shorts beeps and then the
inbound voice quality degrades massively. It sounds like the other user is
a robot...etc.
I'm guessing something (aastra 480 or Asterisk 1.4) is warning
Hi folks,
If you are running a call centre (large or small) using Asterisk, I'd be
interested to know how you log your agents in & out:
E.g.
- Do you use AgentLogin (to force calls onto the agents, perhaps)?
- Do you still use AgentCallbackLogin?
- If you use AddQueueMember, are you
- ru
Greetings,
The Asterisk.org development team has released Asterisk 1.4.19-rc1. This is a
test release for 1.4.19. The official 1.4.19 release will be made after a
1.4.19 release candidate goes through a few days of testing without finding any
major regressions.
This release is available for dow
Greetings,
The Asterisk.org development team has released Asterisk 1.6.0-beta5. As of this
beta of 1.6.0, 1.6.0 is now feature frozen. In addition to a number of bug
fixes, the following new features have been added since beta4:
* The SMDI interface in Asterisk has been reworked to fix a numbe
I was going to ask the same thing today as i am looking for better and more
efficient ways to run a call centre using asterisk!
Look forward to some responses.
Kev
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ade Vickers
Sent: Thursday, 6 March 2008
Hi Steve -
I could be mistaken, but I think this has to be done physically from
the phone. I don't think you can do this with central provisioning or
from the web interface.
- Noah
On Wed, Mar 5, 2008 at 3:20 PM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> I setup a number of remote phones on
Hi Bilal -
> I have the following configuration in my iax.conf
> files at asterisk box1 and box2 (two asterisk):
>
> At box1:
> [user1]
>
> disallow=all
> codec=g729
> codec=GSM
>
> At box2:
> [user2]
>
> disallow=all
> codec=g729
> codec=GSM
>
> If G729 is no more available at box1,
Yehavi Bourvine +972-8-9489444 wrote:
> I've found that my Polycom-501 accepts INVITES from any server in the
> world... I would like to restrict it to accept calls only from the servers
> listed in its config file, but I cannot find anything in the documentation.
> Any
> idea?
I don't believe
Hello,
I have many clients(from 10 to 300 seats) running VICIDIAL for call
centers, both inbound and outbound(and blended).
I also have acouple clients that have over 100 agents using Asterisk
Queues for inbound only. One of them wrote a little web page that
integrated with their timeclock applic
Hi all;
I have two asterisk boxes installed in two separated
sites, the internet bandwidth between them is very
good and I am using G729 codec to communicate with
them and IAX.
The problem that side A hears well side B while side B
does not hear well side A !!
I did one thing in side B that in i
Hi All;
Anyone tried to install Asterisk based on UNIX (not
linux)? Which UNIX was good to work with Asterisk?
Regards
Bilal
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/hs
_
bilal ghayyad wrote:
> Hi All;
>
> Anyone tried to install Asterisk based on UNIX (not
> linux)? Which UNIX was good to work with Asterisk?
Linux is UNIX, for intents and purposes related to Asterisk.
Try *BSD.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1
bilal ghayyad wrote:
> Hi all;
>
> I have two asterisk boxes installed in two separated
> sites, the internet bandwidth between them is very
> good and I am using G729 codec to communicate with
> them and IAX.
Try playing around with the adaptive / fixed jitter buffer settings for
IAX2. Also,
> Alex Balashov wrote:
> Linux is UNIX, for intents and purposes related to Asterisk.
Well... not so much! If you want "real" UNIX, go for a BSD or
God forbid, SCO OpenServer.
Their pedigree is from AT&T UNIX (SYS V Rel 4?) which is considered
to be the "real" UNIX. However, as time has gone by
Bill Andersen wrote:
>> Alex Balashov wrote:
>> Linux is UNIX, for intents and purposes related to Asterisk.
>
> Well... not so much! If you want "real" UNIX, go for a BSD or
> God forbid, SCO OpenServer.
>
> Their pedigree is from AT&T UNIX (SYS V Rel 4?) which is considered
> to be the "real"
On Wed, 2008-03-05 at 14:46 -0800, bilal ghayyad wrote:
> Hi All;
>
> Anyone tried to install Asterisk based on UNIX (not
> linux)? Which UNIX was good to work with Asterisk?
>
> Regards
> Bilal
On what ground would you refrain from using any linux based distro?
___
On Wed, Mar 5, 2008 at 6:09 PM, Alex Balashov <[EMAIL PROTECTED]> wrote:
> bilal ghayyad wrote:
>
> > Hi all;
> >
> > I have two asterisk boxes installed in two separated
> > sites, the internet bandwidth between them is very
> > good and I am using G729 codec to communicate with
> > them and
On Wed, Mar 05, 2008 at 05:24:53PM -0600, Bill Andersen wrote:
> > Alex Balashov wrote:
> > Linux is UNIX, for intents and purposes related to Asterisk.
>
> Well... not so much! If you want "real" UNIX, go for a BSD or
> God forbid, SCO OpenServer.
>
> Their pedigree is from AT&T UNIX (SYS V Rel
On Wed, Mar 5, 2008 at 6:53 PM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> On Wed, Mar 5, 2008 at 6:09 PM, Alex Balashov <[EMAIL PROTECTED]> wrote:
> > bilal ghayyad wrote:
> >
> > > Hi all;
> > >
> > > I have two asterisk boxes installed in two separated
> > > sites, the internet bandwidt
And we found (recently) that if you send the right http packet to a snom
phone you can make the screen say "Agent 155" rather than the extension
number. :)
PaulH
On Wed, 2008-03-05 at 17:27 -0500, Matt Florell wrote:
> Hello,
>
> I have many clients(from 10 to 300 seats) running VICIDIAL for c
I have done integrations with sipsak to do this with Snom phones as well:
http://sipsak.org/
MATT---
On 3/5/08, Paul Hales <[EMAIL PROTECTED]> wrote:
>
> And we found (recently) that if you send the right http packet to a snom
> phone you can make the screen say "Agent 155" rather than the exte
I don't have access to an asterisk system right now
(nor any other sort of information source) but I seem
to recall that from 1.4 onwards the config option for
recording queue calls is named differently...
Is it mixmonitor ? Check you 1.4 queues.conf sample.
PS: I'm not really sure ab
That stinks. Terrible design flaw.
It wouldn't be such an issue if the whole Polycom page (or any
indication of device) prior to the authentication.
I bet one could google for Polycom phones and login as Polycom:456.
Not that one could do much more than brick the phone (which is bad
enough). On
Steve Totaro wrote:
> Try using SIP. Post back with results.
Or that. Certainly, my own experiences with IAX2 would support this
conclusion.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 33
I wouldn't know how to do it the way you mention it,
via local channels...
Our implementation performs ringing transfers via
AMI redirect... The user action is performed on the
desktop, not on the ringing phone.
--
exvito
___
-- Bandwidth an
Sipsak is a lot of fun... :)
PaulH
On Wed, 2008-03-05 at 19:18 -0500, Matt Florell wrote:
> I have done integrations with sipsak to do this with Snom phones as well:
> http://sipsak.org/
>
> MATT---
>
> On 3/5/08, Paul Hales <[EMAIL PROTECTED]> wrote:
> >
> > And we found (recently) that if
On Wed, Mar 5, 2008 at 12:41 PM, Raúl Gómez C. <[EMAIL PROTECTED]> wrote:
> Hi list,
>
> I'm wondering if it's possible to transfer a call that is still ringing??? I
> Have some Grandstream GXP-2000 and with the TRNF button it's impossible. So,
> I've configured some keys to transfer the calls like
I just want to confirm couple of things.
a) If I change an entry in sip.cfg (voice.volumne.persist for example)
on the boot server, the only way I could effect the change on the phone
is to reboot it.
b) What about if I set prov.polling.enabled, will the night time poll
effect the change (even if
Hi All,
I've just compiled Asterisk 1.4.18 and I'm planning on using an LDAP tree
where the users will each have their account, SIP username/password,
extension number, context, etc. My first question is: can this be done with
1.4.x? If so, where can I get the res_config_ldap from??
I googled qui
Interesting. Which type of SIP packet did you send?
The method Paul is referring to below is a page you get GET with certain
parameters to reconfigure the phone via HTTP...
Matt Florell wrote:
> I have done integrations with sipsak to do this with Snom phones as well:
> http://sipsak.org/
>
> M
I believe that IAXVAR in Asterisk 1.6 will do what you want. I have a
backport of this for Asterisk 1.2.14 or so floating around somewhere but it
hasn't been maintained or used for months, may not be compatible with the
1.6 implementation and I offer it with no support whatsoever.
Craig
-Ori
Here is the sipsak string I use:
/usr/local/bin/sipsak -M -O desktop -B "MESSAGE HERE" -r 5060 -s
sip:[EMAIL PROTECTED]
Where "MESSAGE HERE" shows up on the Snom phone display after this is sent.
Of course this is just an example, in the application messages are
posted on the phone display dyn
> I could be mistaken, but I think this has to be done physically from
> the phone. I don't think you can do this with central provisioning or
> from the web interface.
As far as I recall it can be done from the config file only. Here is the
relevant line from sip.cfg:
> As far as I recall it can be done from the config file only. Here is
the
> relevant line from sip.cfg:
>
> device.auth.loc
> alAdminPassword="YOUR-PASSWORD-HERE" />
>
What sip release are you referring to?
I am looking at sip 1.6.x and sip.cfg only allows you to set the length
of the user and
Polycoms need to reboot if you do much more than pick up the handset and
dial a number. A change of config of /any/ scale certainly qualifies here.
If you alter the sip.cfg file on your TFTP/FTP/HTTP server, the Polycoms
/should/ pick up the fact that config file has changed and reboot when
it de
Ahh... our method actually set an alias on the phone for the first
identity rather than simply sending a message to the phone. This means
that whenever the phone would have displayed it's extension number or
name, it would display the string we'd set instead. We used this for
roaming extensions r
That sounds great, could you post an example?
MATT---
On 3/5/08, Rob Hillis <[EMAIL PROTECTED]> wrote:
>
> Ahh... our method actually set an alias on the phone for the first identity
> rather than simply sending a message to the phone. This means that whenever
> the phone would have displayed i
> Polycoms need to reboot if you do much more than pick up the handset
and dial a number. A change of config of any scale certainly qualifies
here.
===That is a bit disappointing!
> If you alter the sip.cfg file on your TFTP/FTP/HTTP server, the
Polycoms should pick up the fact that config file h
> Without portfast, you're looking at about 30
> seconds for STP to negotiate whenever the port bounces, during which
> time higher layer protocols are unavailable. This may interfere with
> CDP and DHCP, if you're using those.
I am using DHCP and I could briefly recall my PC "hanging" for a short
bilal ghayyad wrote:
> Anyone tried to install Asterisk based on UNIX (not
> linux)? Which UNIX was good to work with Asterisk?
Works fine for us, in a FreeBSD jail. Very lightweight (100K RAM
and .5G disk) and very secure even without a jailed environment.
Also prefer BSD ports to Linux' RPMs or
Unfortunately the Polycom's propensity to reboot at the drop of the hat
is one the things I really dislike about the phone - especially when
coupled with the fact that they take so /long/ to reboot.
I must admit, I'm surprised that they don't handle the config file
changing for them on the server
Glad it worked for you. The warning is normal for Cisco switches.
Basically, the "portfast" setting disables Spanning Tree (802.1d &
friends) negotiation for the ports. It is only dangerous if you
actually connect that port to another bridging device that could
potentially have an alternate layer-2
Hi again :)
I've downloaded, compiled & installed 1.6.0-beta4 --with-ldap. After a few
hours of messing with it, I've managed to get it to say that it has
connected successfully to the LDAP backend (by looking at the output of
"realtime ldap status").
I've modified extconfig.conf to what it shoul
exten => s,n,TrySystem(wget -qb -O /dev/null -o /dev/null
"http://${new_ipaddr}/dummy.htm?settings=save&user_idle_text1=${ARG1}+(Roam)")
This is a clip from the macro in question. How you obtain the IP
address of the phone very much depends on your own install - be it via
sip.conf or RealTime.
> especially when coupled with the fact that they take so long to
reboot.
It reminds me of the "good" old days of Microsoft Win95 and NT.
> I must admit, I'm surprised that they don't handle the config file
changing for them on the server better - I had thought they were better
than that.
Yes, I j
Hi Raul
Raúl Gómez C. said the following on 05-Mar-08 07:40 PM:
Ian,
I'm unable to transfer calls using *2, I'm not sure why. Here's my
configs:
In the phones the "/Send DTMF:"/ is set to "in-audio" and "via SIP INFO"
It should only be set to SIP INFO, or else the audio comes out too
chop
Hi
I am quiet new to Astrisk PBX. My question is there any way to know
about the message sequence of a conference from Astrisk PBX.
If anybody knows please help me.
Thanking You
Preethi
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On Wed, 5 Mar 2008, Carole Migden wrote:
> Works fine for us, in a FreeBSD jail. Very lightweight (100K RAM
> and .5G disk)
Make that 100M RAM, of which only about 20M is used by Asterisk.
Roger
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Gareth Owen wrote:
>
>You need to configure the proxy address to be the same as the registrar
>address.
>
I tried that, too - no change
After a while I entered the server ip into server fields of the gui and
set all ports to 5060.
In the section troubleshooting I defined the asterisk server
On Thu, Mar 6, 2008 at 5:32 AM, Carole Migden <[EMAIL PROTECTED]> wrote:
> Generally what you know is best
This is close to the best advice I've seen on this list in the last 5
years! The rest is a question of religion ;)
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