If anyone could please help me with this, that would be very helpful.
Miles Scruggs wrote:
Just starting to enjoy the full features of asterisk, I do have a
couple questions though, that I can't seem to find answers for in the
wiki, just wondering if someone could light my way.
after a
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You just add the same agent to both queues (don't use groups), like in
queues.conf:
[queue1]
member=Agent/101
[queue2]
...
member=Agent/101
Now Agent 101 is a member of both queues, and will not be called while
s/he
We have a SNOM360 (ext 226) configured for redirection (away on annual leave) to another SNOM360 (ext 225), being tested from a SNOM320 (ext 227) which appears on the surface to be an easy adjustment.Was receiving the following message, "Got SIP response 302 "Moved Temporarily" back from"of
I am not sure if it has been fixed, but using groups used to behave
erratically in earlier versions of *, so I am not used to doing it. A few
more configf lines are well worth the added stability.
l.
On Thu, 30 Mar 2006 10:30:50 +0200, Tomislav Parčina [EMAIL PROTECTED]
wrote:
In
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between the DMZ and our LAN.
Also
exten =
Hello,
On Mar/30/2006, Fran wrote:
There are 2 ways in spain to subscribe SMS with a fixed-device (with
Telefónica)
-Activation Button: The device send an administrative message to subscribe
SMS. There other administrative messages such as: unsubscribe, list
managing, send to a list,
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi all,
I would like to hear from you, SjPhone has the option to Do not Send
silence (audio options, advanced), should i use this or remove this
option?
Everything ran well until now, but there was few people on my server,
i'm
I guess Protocol 1 is UBS1. I think it should be.
No, i have never tested Asterisk sending messages.
We have tested some fixed devices (UBS1, UBS2 Domo type)
The UBS1 SMS Service is 900716800
What error do u have? Timeouts? etc?
greetings,
Fran
-Mensaje original-
De: [EMAIL PROTECTED]
I want to
replace a Telebutler software auto attendent system that used a 4 port Dialogic
board connected to a Panasonic KXTD 1232 6 line system. We have spare computers
here. How do I connect asterisk to this Panasonic system?
___
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In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
The h323 channels doesn't have any support for NAT. You'd need to
register with a properly configured gnugk for that.
E didn't mention NAT, he only spoke about firewall.
--
Tomislav Parcina
tparcina#lama.hr
HI:
Is there any way to raise up sound volume on fxo on
TDM04B without changing tx-gain and rx-gain ?
__
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jonny hashem wrote:
HI:
Is there any way to raise up sound volume on fxo on
TDM04B without changing tx-gain and rx-gain ?
No.
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Hi,
I compiled and installed Asterisk and mISDN yesterday. It worked after
some errors. There's one HFC-Card which should be in nt-mode, and which
worked yeterday with my phone here, and there's another isdn-card with a
winbond chip i think, but i don't want to use this one at the moment.
Hi all,
Can i make Asterisk stay out of the media path forcall park feature?In the 'sip.conf' i made canreinvite=yes in the general sectionbut it does not seem to take effect. I don't see any reason for Asterisk to withhold sending re-invite. I am testing the call park in the single LAN,both on
On 3/30/06, Neil Skowronek [EMAIL PROTECTED] wrote:
Hi list,
Havent seen any recent postings regarding Voice Recognition
I checked out Sphinx a few months ago but was wondering if anyone has had
any
recent experiences, success/failure stories with any new or updated Open
Source VR software
Also have the same trouble connecting Asterisk with the Huawei
softX3000 softswitch via SIP.
Anyone have any experience with the Huawei switch. I am only in
control of the Asterisk server, Huawei softX3000 is controlled by
another company.
I have included the ouput from the SJPHONE which can
Hi
As far as I know you shouldn't use hisax AND mISDN together. Especialy not
configured for the same card.
Try disabling the hisax driver in your kernel and recompile.
hisax 442192 0
crc_ccitt 1952 1 hisax
isdn 119456 1 hisax
slhc
OK - I should have been more specific in my original post, but before I get too
excited to try this tonight when I get home, does this work with ANY vonage
line (ie: first, unlimited line) or does it have to be on a softphone enabled
line?
Thanks!!
From:
On 3/29/06, Nathan Bowyer [EMAIL PROTECTED] wrote:
When I look at the code, in this case calldate is actually thecdr-start value.I'm working on a patch to record answer and end as
well.
Thanks Nathan. Are you going to post this to the bugtracker?
-Brian
Is this only for the secondary Vonage SIP account, or does this work in
place of the Vonage supplied ATA?
John Novack
Adrian A wrote:
Works great with these settings:
[vonage]
type=peer
secret=password
username=phone number
host=sphone.vopr.vonage.net http://sphone.vopr.vonage.net
I have a problem with audio and hope that somebody can give me some
hints how to track it down and eventually solve it:
I got 4 phones, lets give them numbers:
100 and 200 are in my office, they can reach each other without problems
300 and 400 are outside of my office, each on a different
Any opensource solution?Thanks,WaldoOn Mar 29, 2006, at 7:25 PM, Alyed Tzompa wrote: I use Portaone's PortaSIP for everything related to LCRAlyed Return-Path: [EMAIL PROTECTED] Wed Mar 29 16:48:54 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
The lookupblacklist cmd, looks to me like it only handles inbound numbers.
Can I use this command just to block outgoing numbers, by where I place it
in my outoging call settings?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Melcon Moraes
Sent:
I am not getting any caller Id with my standard T-1. Is a
standard T capable of sending callerid? I dont want to
spend time troubleshooting my PBX if Asterisk cant send it down that
type of trunk.
Jordan
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This is not a dumb question.
Most of the other replies I have read mentioned various ways to connect
to the pstn. I wanted to mention why it makes sense to do that. Many of
the companies I have installed asterisk for didn't even have their
system on a network with a gateway. They have dedicated
In article [EMAIL PROTECTED],
Jim Houser [EMAIL PROTECTED] wrote:
I wanted the user interface of FreePBX over what is provided in the latest
version of AAH. I installed the latest version of AAH and then just
installed FreePBX over the top. It went fantastic and I do like the FreePBX
web
I think there is a bug related to this. I haven't been able to track it
down or really recreate it with any certainty yet. When I do I'll post
something to Mantis. If you have any info to share with me about your
situation when this occurs let me know.
I have noticed that I can get it to occur
Yes, It can send Caller ID down the T1 line..
Some T1's only accept Caller ID's that match the set of DID's associated
with the T1.
Others, like mine, will take anything you send it...
Chad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jordan
NovakSent: March 30, 2006
Jordan Novak wrote:
I am not getting any caller Id with my standard T-1. Is a standard “T”
capable of sending callerid? I don’t want to spend time troubleshooting
my PBX if Asterisk can’t send it down that type of trunk.
Its kind of hard to figure out exactly what you're asking based on the
Hi,
maybe a dumb question, but it seems that some calls are directed to our
central dial in number despite the extensions the callers say they dialled.
E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown
extension, where it is right, and redirects the call to the
Melcon Moraes wrote:
What PBX is that? What do you see at CLI when you call that port?
Actually, this behavior exists in the PBX. I need to make asterisk works like
a fxs that will receive the calls, and then route them in two diferents SIP.
Using your example:
I call to the extension 100 or
If your T1 is ISDN, you should automatically have it. If
your T1 is RobBit, you have to check with the CO to see is the ANI/DNIS service
is turned on (it is seperate service for RobBit T1), and the T1 is usually set
to wink start.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I have found this to be true also.
[whatever] has to match username=
It appears that it ignores the username field for IAX users.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tomas Komarek
Sent: Monday, March 27,
Greetings, All-Knowing Asterisk Users List,
My company needs to build a reliable fax server that can handle at
least 30 simultaneous incoming faxes from the PSTN, using PRI. We
realize that this can be solved in any number of ways using a Linux box,
but since IVR is also a must,
I'm so confused right now. Asterisk is letting me log into queues
with agent ids that do not exist in agents.conf, it is not prompting
me for passwords... any thoughts? I am running 1.2.5
On 3/30/06, Lenz [EMAIL PROTECTED] wrote:
I am not sure if it has been fixed, but using groups used to
Sharath Chandra wrote:
Hi all,
Can i make Asterisk stay out of the media path for call park feature?
No.
When a call is parked, it's parked on the Asterisk system, so it will
stay in the media path.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
Miles Scruggs wrote:
Just starting to enjoy the full features of asterisk, I do have a
couple questions though, that I can't seem to find answers for in the
wiki, just wondering if someone could light my way.
after a caller has made their choice of options in the dial plan, I
would like them
I'm using Asterisk a SIP Server for a lot of GrandStream HandyTone
ATA's. Each one of them is configured in sip.conf as:
[1234567]
type=friend
username=1234567
secret=1234567
callerid=ATA 1234567
host=dynamic
nat=yes
qualify=yes
disallow=all
allow=g729
canreinvite is set globally to YES.
When
We have set it up on an 81C and an 11C using PRI as the interface. The
only issue we've run into is that the 81C shows the full 10-digit caller
ID value instead of the 4-digit extension. We don't have that issue on
the 11C for some reason. It may have to do with having networking
enabled (so we
Josué,
My setup is:
txfax -[Asterisk] - E1 Pri -[Cisco 5350]- t.38 with Sip - [Terminating
Carrier]
So the t.38 is all done by the Cisco 5350 and not by asterisk at all.
I have done some tests with a Cisco ATA186, but with limited success using
g.711 pass through.
I can receive faxes
Hi,
Does Asterisk have
builtin (T1 or E1) span monitoring? If a span goes down, will asterisk know
about it. Personally, I would like to have a event generated through the Manager
API interface.
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Hello,
I wish to play a recorded announcement on reply to 4xx
5xx 6xx messages .
According to the status a audio file would be played
from asterisk server via ser to the caller
How can I configure a such feature ?
My configuration:
Ser act as an outbound sip proxy .
Asterisk a sip media
Hello,
I bought a Grandstream Handytone 486 to forward incoming calls from our old
analogue PBX to the asterisk server.
My first test was connecting an analogue phone to the Handytone and calling a
sip phone - worked.
Now I used the same cable to connect the line port of the Handytone to the
Wai Wu wrote:
Does Asterisk have builtin (T1 or E1) span monitoring? If a span goes
down, will asterisk know about it. Personally, I would like to have a
event generated through the Manager API interface.
Have you actually tried this? It takes all of about 10 seconds to answer
this question
Is there anyway for me to authinicate first before givng options
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JP Carballo
Sent: Wednesday, March 29, 2006 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
This is for the softphone account. I tried one for about a month just to
see how well it works with asterisk. A few notes(from memory):
1) Seemed reliability and quality were better than other retail
providers I tried.
2) Tested it with 2 channels open from pstn calls and then opened a
third by
Hello,
On Mar/30/2006, Fran wrote:
I guess Protocol 1 is UBS1. I think it should be.
ok, me too...
No, i have never tested Asterisk sending messages.
We have tested some fixed devices (UBS1, UBS2 Domo type)
I have only checked Domo phone, but I don't know which protocol it is using.
You MUST have a softphone account. Some blog said they may open it up to
all accounts at some point in the future but that is only a rumor right now.
-Original Message-
From: Steve Jones [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 30, 2006 4:58 AM
To: Asterisk Users Mailing
Kevin P. Fleming wrote:
Wai Wu wrote:
Does Asterisk have builtin (T1 or E1) span monitoring? If a span goes
down, will asterisk know about it. Personally, I would like to have a
event generated through the Manager API interface.
Have you actually tried this? It takes all of about 10
Sorry to asked the question here. Just found out that it actually generate an
event. My T1 is being installed right now, just to want to what to expect.
From: [EMAIL PROTECTED] on behalf of Kevin P. Fleming
Sent: Thu 3/30/2006 10:57 AM
To: Asterisk Users Mailing
Hi
I am looking at purchasing some DID lines from Teliax to install it on my asterisk.
i would like to know some feed back on Teliax before i purchase.
suggest me if there are better sevice providers.
thanks
Giridhar Bandi
___
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Hi. We have an odd problem with incoming SIP calls. I have attached a
SIP debug log, with some asterisk verbosity as well, demonstrating the
problem, below.
Is this a known bug?
Vital stats:
- Asterisk 1.2.3
- Sipura SPA-841, SPA-941 phones
- Fedora core 3
The problem manifests itself with
Hi all,
I've my Server running well, then sometimes Sjphones looses registry
and it only works well again if i restart the pc running sjphone.
Has any one experience this?
Best regards,
Marco Mouta
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Hello,
one of our clients is currently testing three installations:
- Cisco Callmanager 5
- Siemens HiPath 8000
- Asterisk
To get an impression how these system behave under heavy load, he's going to
use an ABACUS 5000 system
That mean if I can't patch * if a flaky lines are deleted, I can't
reroute the call (using off-B channel transfer) to a different span.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: Thursday, March 30, 2006 11:13 AM
To: Asterisk
When I was in Telefonica I+D I developed an software for windows that allows
sending sms throw an ISDN line. It was more than 3 years ago and I don't recall
to many details but we had to implement ETSI ES 201 912 and
make an V28 modem emulation over ISDN.
-Original Message-
From:
When it works it works great. We have had a few issues
lately but they were resolved fairly quickly.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giridhar
Reddy BandiSent: Thursday, March 30, 2006 8:14 AMTo:
asterisk-users@lists.digium.comSubject:
Hi,
I am not sure if this is a bug in FOP (Flash Operator Panel), a
configuration error on my part or a bug in Asterisk. I am using Asterisk
1.2.5 and Zaptel 1.2.4 on a Centos 4.2 server with Linux version
2.6.9-22-EL-i686. Kernel updates are excluded and the server has been
updated using
Hi,
What does your SIP config look like for the SJPhone? Also what operating
system does this PC have and is it up to date with security and bug patches.
Thanks
Marco Mouta wrote:
Hi all,
I've my Server running well, then sometimes Sjphones looses registry
and it only works well again if i
Dear Group;
I am closer to where I want to be. I could still do with some help.
For my Internal(*)I setup the following;
extensions.conf
---
[SIPOUT]
exten = _6.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
If I dial sip:[EMAIL PROTECTED] I see the call go to the External(*)
In my
hi Ralf,
AFAIK, the 486 CANNOT do what you want it to do.
you need a call thru functionality and this can be achieved only with 488 .
then you conect the line to the old analog PBX extension and any call coming to this extension will enter the Asterisk as a call from a SIP extension.
i hope i
Can you post your dialplan? We'd be much better at troubleshooting the
problem if we could follow the path that calls take.
Aaron
On Thu, 30 Mar 2006, Sebastian Reitenbach wrote:
Hi,
maybe a dumb question, but it seems that some calls are directed to our
central dial in number despite the
Hello,
I have an * server configured with a voip provider
and an ISDN BRI backup line.
When Internet is down, internal SIP calls or ISDN calls
fails.
I have analysed some internal sip traffic and
asterisk doesnt answer fast enough to phones.
It sends DNS queries to find voip provider
Hi Adolfo,
I have done this and it works. I have maxed out an E1 with 30 concurrent
calls of which at least 25 would have been fax.
Hardware is nothing special, Dell Poweredge 750, 512mb ram, single SATA
drive with either of a TE410p or TE110p card. OS is FC2 with kernel 2.6.9
I expect
Windows XP service Pack 2
What you mean with SIP config look like?
I've everything by default, only config for Calls through SIP proxy
Bug patches from sjphone?
On 3/30/06, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
What does your SIP config look like for the SJPhone? Also what operating
I am happy teliax user,
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Giridhar Reddy Bandi wrote:
Hi
I am looking at purchasing some DID lines from Teliax to install it on
my asterisk.
i would like to know some feed back on Teliax before i purchase.
Teliax has worked great for me for the past year. There were a few
issues at the beginning, but a call to
Hi all,
I would like to point out a problem I observed.
I installed a new asterisk server, very similar to another. So, after
complete installation
of asterisk and AMP, i tried to import back a ful AMPl backup from the
first AMP / Asterisk box
Everything was (quite) OK. the only problem was that
Adolfo R. Brandes wrote:
After combing these lists for clues, we began experimenting
extensively with Asterisk and its software DSP and fax capabilities in
most of their incarnations, such as Rxfax or Iaxmodem/Hylafax,
together with Digium's E1 cards in server-grade Intel motherboards,
Hi there,
I am writing some Ruby (Rails) app which needs to comunicate some
events straight to phones.
The application runs in one box.
Asterisk runs in another. (I am already able to config a basic *)
My questions are:
Where do I start learning how to send SIP messages to a phone?
any
Ok, this is highly confusing.
hestia*CLI sip show users
Username Secret Accountcode Def.Context
ACL NAT
2944030 2944030 oneeighty_start
No RFC3581
2944035 2944035
I contacted the developer of irqbalance (Eric Dorland) and he responded
as follows:
Certainly it can't cause any serious problems, just performance ones.
And yes, it is my understanding that later 2.6 can do irq balancing
themselves if the right option is set when compiling it.
Matthew Roth
Something I've been curious about is if it is possible to stick their
ata on a extra ethernet port on an Asterisk server and have the Asterisk
server spoof the Vonage server. Then, do a man-in-the-middle type thing
to use the ata for authentication, but have Asterisk handle all the calls.
Check out sipsak. http://sipsak.org/
On 3/30/06, Andres Paglayan [EMAIL PROTECTED] wrote:
Hi there,
I am writing some Ruby (Rails) app which needs to comunicate some
events straight to phones.
The application runs in one box.
Asterisk runs in another. (I am already able to config a basic
Hi,
If your sip.conf is not setup properly SJPhone will not work. Here is my
SJPHpne SIP config:
sip.conf**
...
;SJphone
[410]
context=longdistance
;canreinvite=no
type=friend
username=410
secret=passwd410
callerid=410
qualify=yes
nat=no
host=dynamic
[EMAIL PROTECTED]
disallow=all
I contacted the developer of irqbalance (Eric Dorland) and he responded
as follows:
Certainly it can't cause any serious problems, just performance ones.
And yes, it is my understanding that later 2.6 can do irq balancing
themselves if the right option is set when compiling it.
Matthew Roth
This is getting very annoying.
Thought it might be a irq conflict/sharing issue -- so resolved that.
Still, cannot get an incoming call to bridge with IAX or ZAP devices.
qozap: dropped audio. errors remain as well
Setup is in Germany -- TE mode,
Someone plase help
This is terrible.
Craig Guy wrote:
I have done this and it works. I have maxed out an E1 with 30
concurrent calls of which at least 25 would have been fax.
Thank you for your prompt reply! Now it's clear to me that at least it
can be done! I'll try replicating your scenario to see if I can also
get good
Lee Howard wrote:
However, based on the comments you give I'd suspect that you're having
what people seem to be calling frame slipping. There seem to be some
motherboards that react poorly with Zap cards (or their respective
drivers) and cause that. Your zttest results should be revealing
OH ya, maybe some console output would clear this up for you:
Call comes in, goes into incoming call context...:
-- Executing NoOp(Zap/1-1, 49CALLERID) in new stack
-- Executing NoOp(Zap/1-1, 0) in new stack
-- Executing GotoIf(Zap/1-1, 0?s|4:s|5) in new stack
-- Goto
Adolfo R. Brandes wrote:
Lee Howard wrote:
The concurrent calls really isn't that big of a deal, either, if
those are your thoughts. The bigger issue seems to be the quality of
the audio as it is delivered to the fax application/modem.
Interesting. The little information I've found
Looked around a little. If you set nat=never, then it won't set the
phone to RFC3581... I haven't tested it, but you may want to try it :)
Aaron
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
Nat=
-This option determines the type of setting for users trying to
You could have something parse the output of /proc/zaptel/1 (depending on which
card you have you may have additional files there). If there is an alarm it
should be displayed there. This assumes you are using zaptel though.
--johann
Kevin P. Fleming wrote:
Wai Wu wrote:
Does Asterisk
On Thu, 30 Mar 2006, Giridhar Reddy Bandi wrote:
I am looking at purchasing some DID lines from Teliax to install it on my
asterisk.
i would like to know some feed back on Teliax before i purchase.
suggest me if there are better sevice providers.
I have had issues with termination on teliax.
Try LCDial() command. YOu can find it on the
Wiki. It uses an SQL backend and it works great.
- Original Message -
From:
Waldo
Rubinstein
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Sent: Thursday, March 30, 2006 5:14
AM
Subject: Re:
Is there any way to get Asterisk to reload the /var/lib/asterisk/astdb file?
It seems to only read it on startup.
Thanks.
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Adolfo R. Brandes wrote:
Lee Howard wrote:
However, based on the comments you give I'd suspect that you're having
what people seem to be calling frame slipping. There seem to be
some motherboards that react poorly with Zap cards (or their
respective drivers) and cause that. Your zttest
Douglas Garstang wrote:
Is there any way to get Asterisk to reload the /var/lib/asterisk/astdb file?
It seems to only read it on startup.
Thanks.
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Don Pobanz wrote:
Adolfo R. Brandes wrote:
Lee Howard wrote:
However, based on the comments you give I'd suspect that you're
having what people seem to be calling frame slipping. There seem
to be some motherboards that react poorly with Zap cards (or their
respective drivers) and cause
Hi,
This problem may be related to a configuration problem but I believe it
is a bug in the FOP since restarting the FOP server clears the problem.
Here is the scenario: Using AgentCallBackLogin and have four agents
logged in a call is made to one of the agents directly from an internal
How do you disable call waiting on Polycom IP601 phones?
I've looked through the user and admin guides and can't see anything about
disabling it.
-Dan
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Hello all,
Is there a way to check who is using the VAD option whenever
we get the message VAD frame at the end at the CLI ?
--- The IP is not listed.
Some people use it but I can't tell them to turn it off for better audio
performance,
and I know it generates a lot of messages on the
Hi All,
We are offering Romania Bucharest DIDs
NXX : 4021+ and 4031+
We have plenty available on http://www.didx.org/
Regards,
Oliver
Vermeulen
World Venture
Group Telecom
Tech
/ Admin
Corporate Address:
Str Avionului Nr 35/bl16J/3
Bucharest, 014333 Romania
In polycom's firmware version 1.5.x, it's in the phone menu:
Menu -
3 (Settings) -
2 (Advanced) - (Enter Admin Password, '456' typically) -
1 (Admin Settings)-
2 (SIP Configuration) -
Line 1 (or 2 or 3 or ...) -
Calls per Line Key, Set it to 1.
Prior to version 1.5.x, it must be implemented in
I have an odd problem between a Nortel BCM and Asterisk (1.2.4) Zaptel (1.2.3).Calls all appear to be working fine, however, when someone originates a call from the PBX and the Switch returns a 486 (Busy) the PRI channel hangs in a busy state indefinitely. Subsequent to this, the PBX side shows
exten = s,2,Authenticate(/home/me/calls|[|a])
I use the above in my dial plane to auth users, up untill now it worked
fine. All of a sudden it just stopped working. Any ideas of why I can no
logerlog in with the numbers contained within
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On Thu, 30 Mar 2006, Mojo with Horan Company, LLC wrote:
In polycom's firmware version 1.5.x, it's in the phone menu:
Menu -
3 (Settings) -
2 (Advanced) - (Enter Admin Password, '456' typically) -
1 (Admin Settings)-
2 (SIP Configuration) -
Line 1 (or 2 or 3 or ...) -
Calls per Line Key, Set
Joshua,
I'd like issue a command, that would cause Asterisk to re-read it, and refresh
what it has in memory.
Douglas.
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 30, 2006 12:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Joshua Colp wrote:
It depends what you mean by reload. The file is opened when Asterisk
is started, and then anything can use the astdb. How it's used is up to
what is using it. You need to be more descriptive. What data do you need
to reload from the astdb?
And even more importantly...
If all you want to do is not play a tone to the users if they are on
the phone, then modify the tone in sip.cfg or phon1.cfg, I don't
remember which one it is, but I have done it before.
On 3/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Thu, 30 Mar 2006, Mojo with Horan Company, LLC
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