On 2 Apr 2006, at 04:27, Rich Adamson wrote:
Kevin P. Fleming wrote:
Rich Adamson wrote:
Is this worthy of opening a bug assuming the above comment is still
valid? Would the individual(s) maintaining res_snmp want to log
into
either of these internet accessible boxes to identify the
I have a asterisk server running on site listening on a public ip.
Tonight I attempted to connect a Cisco 7960 phone from my home location via sip
but failed. My home network is simple, Cox cable connection hooked to a
linksyswrt router. The firewall on the linksys router is disabled
and I
This is my debugtried with netmeeting I can still hear but when I talk nothing happenmygw*CLI h.323 debugH323 debug enabledmygw== New H.323 Connection created. -- Setting up Call -- oCall token: [ip$192.168.1.219:1057/8226]
-- oCalling party name: [myPersonal] -- oCalling party number: [] --
On 4/2/06, Christian Gröger [EMAIL PROTECTED] wrote:
need that Zaptel stuff? It always prompted errors so i am now using
mISDN -without errors, is there a module for freePBX for mISDN?
to use mISDN with freepbx, you can Add custom trunk in the Trunks menu.
Anyway, is there a good manual for
I am not sure if this debug message is enough information.
Try to do what I told. Switch to another H323 channel driver and see what
happens. Try first chan_oh323.
Michael Mansos(or something like that) and other guys have been done a
good job.
Isamar
Could anyone shed some light on this problem:
Running Asterisk @ Home 1.7
When call comes in (Zap Clone 100 Card), the extensions ring but when
lifting the receiver on an IP phone (BudgeTone 100), I just hear a bit of
crackle sound in time with the ringing of the other extensions. The external
Is there any way to get a polycom 601 to do overlap dialing?
-Dan
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Is there any way to get a polycom 601 to do overlap dialing?
I can't find anything on the subject, which confirms my initial hunch:
I really doubt it. You could probably work something up in asterisk,
though.
- Noah
___
--Bandwidth and Colocation
Hi Again Avi -
Sadly, that doesn't work -- the Polycoms store their
directories locally as well and re-upload them on reboot.
Another idea: Can you create the mac address-directory.xml files as
symlinks to the central file? Maybe if the phone sees a directory
file already there it will not
Hi Sam -
Thanks for your link. how to build asterisk into
this hardware?
As mentioned earlier, have a look at astlinux:
http://www.astlinux.org/
There are pre-built versions for soekris/wrap, and general x86
computers. Kristian (the astlinux developer) made this run on a
gumstix, too, but I
Hi Bjorn -
Everything you mentioned seems to point to the problem being a
hardware issue, or more specifically the way that FC and CentOS are
using your hardware.
Why not use different hardware and/or OS? Maybe FC and CentOS just
use faulty driver for your NIC?
- Noah
Shaun wrote:
I have a asterisk server running on site listening on a public ip.
Tonight I attempted to connect a Cisco 7960 phone from my home
location via sip but failed. My home network is simple, Cox cable
connection hooked to a linksys wrt router. The firewall on the
linksys router is
Or, could he be seeing an outstanding issue with counting (eg, start
with 0 or 1)?
Seems like that might be the case. I've got about ten g729 licenses and
never see any warning messages, but then again this is a small system
and I don't think I could consume all of them if I tried.
Alyed
end-to-end path. Each step through the tracert process does nothing
more then issue an icmp echo request, measuring the response time and
displaying it.
maybe on windows it does icmp echo but no unix does this (at least not
by default). i recommend you study what unix traceroute actually
Rich Adamson wrote:
Is this worthy of opening a bug assuming the above comment is still
valid? Would the individual(s) maintaining res_snmp want to log into
either of these internet accessible boxes to identify the root cause?
The module loader in trunk is undergoing changes that will
Shaun wrote:
I have a asterisk server running on site listening on a public ip.
Tonight I attempted to connect a Cisco 7960 phone from my home location
via sip but failed. My home network is simple, Cox cable connection
hooked to a linksys wrt router. The firewall on the linksys router is
Hello,
As I know from my experience with Chan323 and OH323.
I setup Asterisk on Redhat 9.0 i386 and it is working without any problem
with Chan323, OH323 libraries required.
I never tried OOH323 come (0.4) with Asterisk. If possible I would like to
know how to use newest version of OOH323
Evar,I tried to duplicate this on my system but wasn't able to. Do you have the same password set for both voicemail boxes? Have you already logged into both of them individually (maybe it's a cookie thing?)
KyleOn 4/1/06, Ever Zalazar [EMAIL PROTECTED] wrote:
Hi everybody..I have the
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the sip
code over the last year)
SVN-branch-1.2-r9156
Have you looked at the sample configs in /usr/src/asterisk/configs?
Yes I have and my own configs are pretty much copies of them.
They do not detail, do
HII have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ?i generally look at the /var/log/asterisk/full
suggest me if there are better way of doing this thanksGiridhar Bandi
Giridhar Reddy Bandi wrote:
HI
I have two sip accounts from two different ITSP's both configured on
asterisk server. how can i know if these accounts have been successfully
registered ?
i generally look at the /var/log/asterisk/full
suggest me if there are better way of doing this
How
Try sip show registry from the asterisk console.Kyle
On 4/2/06, Giridhar Reddy Bandi [EMAIL PROTECTED] wrote:
HII have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ?i generally look at the
Is the SIP phone behind NAT? That's one of the common reasons for one way audio. You might want to try forwarding some port ranges if you are behind NAT just to eliminate that as a possiblity. The SIP port ranges should be something like:
SIP: 5060-5061RTP: 1-2KyleOn 4/1/06, Il Neofita
Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download.ThanksTin Trung NguyenTechnical TeamMobile: 084-91.365.4857website: www.daivietcontrol.net___
On 02/04/06, Nguyen Trung Tin [EMAIL PROTECTED] wrote:
Hello All
I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on
windows, any body could be mail or send to me URL to download.
That version is a year and a day old now, isn't it?
Peter
--
Peter Bowyer
Email: [EMAIL
Nguyen Trung Tin wrote:
Hello All
I read in www.sineapps.com http://www.sineapps.com have Asterisk 2.0
rewritten C# and run on windows, any body could be mail or send to me
URL to download.
Thanks
Tin Trung Nguyen
Technical Team
Mobile: 084-91.365.4857
website: www.daivietcontrol.net
I would also like to know how to do this, it really defeats the whole
purpose of the list if you reply off list.
Please post that to the list.
Miles
Shaun wrote:
Sent you a email
~Shaun
Tom Vile [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
I have a script that will do
On Sun, Apr 02, 2006 at 09:32:09AM +1000, RumaTech wrote:
Sorry, I was out of action for some time.
I am using Voipstnt to plae calls to USA/Canada and I bouhj 1 copy og G.729.
This was mainly to get one of the local Australians VoIP providers working.
Each channel needs TWO licenses, one for
Kristian Kielhofner wrote:
That was last years April Fools joke. I still haven't heard this
years...
And how much do you want to bet that it'll be this years as well.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
Hi,
okay, i managed to install it :) but I have some Problems with mISDN. I
have set up two mISDN-extensions, that can phone each other, or the
mailboxes. But i can't phone those special numbers like *60 for weather
or *98 for message center, what should i do?
stoffell wrote:
On 4/2/06,
Hello all,
I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until
recently all was good. on Friday I was running 1.2.5 when I added the fourth
phone. I have to admit to initially wiring the rj11(crossed wires) wrong the
first time but other than that nothing I can think of.
--- Luis herrera [EMAIL PROTECTED] wrote:
Hi. I have a [EMAIL PROTECTED] setup at my home. My problems is
with
phones outside my network. I call the extensions
without a problem, it rings but when they answer I
can't hear them and they can hear me.
I set up in the SIP.CONF
nat=yes
I'm
Steve Kennedy wrote:
Each channel needs TWO licenses, one for each way (I think).
Nope. The encoder/decoder licenses are counted separately, and each
license you purchase entitles you to one encoder and one decoder.
___
--Bandwidth and Colocation
My guess (it's only a hunch) as to why he asked for
off list is for $$$. Usually when you see that the
person wants $$$. I know there is a way of doing it in
asterisk. My friend has it. Too lazy to get the code
from his box. When I do I will post it. (I also
believe that it is on the wiki).
Dovid
I have searched high and low on the wiki, the problem is that it can fit
under a number of different names, since this process can be used for
tons of different applications. If you find it let me know, I really
want this info, as it is the last piece of the puzzle for my asterisk box.
Your just one day too late. Some one did the same last
year. I have the email some where on my server. Goto
be on the game and original.
--- Nguyen Trung Tin [EMAIL PROTECTED] wrote:
Hello All
I read in www.sineapps.com have Asterisk 2.0
rewritten C# and run on windows, any
Hi Tzafrir, thank´s for your help.
My configurations:
#zaptel.confspan=1,2,0,cas,hdb3cas=1-31:1101loadzone=usdefaultzone=us
span=2,1,0,ccs,hdb3bchan=32-46dchan=47bchan=48-62loadzone=usdefaultzone=us
#zapata.conf[trunkgroup]
--- Thorben Jensen [EMAIL PROTECTED] wrote:
http://voip-info.org/wiki/view/Easy+PABX
With Easy PABX you can create your own virtual PABX
online in just minutes.
Easy PABX is based on Asterisk and best of all -
it's completely free.
Regards
thorben.dk
Whats the catch ? When do you
You can do the following.
Create a file called blacklist.conf . In it create the
following.
[BlackList]
Exten = 5551212,1,Playback(num-blacklist)
Exten = 5551212,2,Hangup
Exten = 1900.,1,Playback(num-blacklist)
Exten = 1900.,2,Hnagup
You get the idea.
In your extensions.conf put in the top
Think people will fall for it again next year too?
Hello All
I read in www.sineapps.com have Asterisk 2.0 rewritten C#
and run on
windows, any body could be mail or send to me URL to download.
Thanks
Tin Trung Nguyen
Technical Team
Mobile:
Think people will fall for it again next year too?
Newbies will
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
___
--Bandwidth and Colocation
I will post this info to the list when I get a chance.
--- Miles Scruggs [EMAIL PROTECTED] wrote:
Hey,
I would like in the course of dial plan logic, to
trigger a separate
outbound call. If that outbound call is answered,
and if that certain
key response is detected then it will
I dont have that option in my phone, this is software 8.2 (p003-08-02 if i
remember correctly)
~Shaun
Doug Lytle [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Shaun wrote:
I have a asterisk server running on site listening on a public ip.
Tonight I attempted to connect a Cisco
Dont have that option, i think i remember seeing it in version 7.xx but i
cant remember, i may have to downgrade it i guess
~Shaun
Rich Adamson [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Shaun wrote:
I have a asterisk server running on site listening on a public ip.
Hi list
automon now works as Monitor does.
But MixMonitor is a better way for most cases, I guess.
Any workaround to make automon do that?
Any help will be appreciated.
Franz Wu
___
--Bandwidth and Colocation provided by Easynews.com --
This is possibly a dumb question, but I've googled around and poked through
the documentation and I'm a bit confused.
My initial experiment with Asterisk involves setting it up in place of my old
dedicated answering machine. That means I've still got a regular old phone
on the line which we
Due to an error in the configuration of the mirroring tool we are using
to mirror the repositories from our internal commit server to the public
read-only mirror, the revision numbers were not being properly kept in
sync (so rev 14381 on the internal server was not the same as on the
mirror).
Shaun Reitan wrote:
I dont have that option in my phone, this is software 8.2 (p003-08-02 if i
remember correctly)
That is the firmware that I am also running.
It's under SIP Configuration, all the way at the bottom. Not within the
extension confgs, keep hitting the down arrow until you
Hi,I am trying to connect Asterisk to traditional central. It must be done over ISDN. I am utilizing Asterisk 1.0.9 from bristuff RC8p. I was able to communicate them with each other, but I have a problem with callerid. The traditional central does not recognize the callerid from the phones
Hi,
I'm not an expert, but as far as i know, your incoming calls will
arrive with DID in ${EXTEN}
so the only thing you need is:
exten = 1234,1,GoTo(context1,1234,1) ; example for context extension
and priority
exten = 2345,1,GoTo(context2,2345,1)
exten = 3456,1,GoTo(context3,3456,1)
Be sure
Here my script:
exten = 230,1,Answer exten
= 230,2,NoOpexten =
230,3,Voicemail(u${EXTEN})exten = 230,4,NoOp(Need results
from VoiceMail() above - should be non-zero and provided to ARG3)
exten = 230,5,NoOpexten =
230,6,GoToIf($[${ARG3} = 0]?s|8) exten =
Use a set callerid before your dial statement in extensions.conf.
-Original Message-
From: Andrew Nowrot [mailto:[EMAIL PROTECTED]
Sent: Sun 4/2/2006 4:19 PM
To: asterisk-users@lists.digium.com
Cc:
Subject: [Asterisk-Users] Connecting
Hi,
I wonder if VoIP providers consume two licenses when one calls via them?
One license for my phone to the provider and one license when call is
passed to the recepient.
Is that possible?
Rudolf
On 4/3/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Steve Kennedy wrote:
Each channel needs
I have the license for G729, however I need to use a different codec for the prepaid service, but when the call is started I have this errorAsked to transmit frame type 256, while native formats is 4 (read/write = 4/4)
___
--Bandwidth and Colocation
Hi,
I have searched the web and found some basic information about the
LOCAL/ channel. I am wondering if anybody has any good web resources
they can point me to on this subject as I think I will need to use it in
the near future in a solution I am putting together.
Alternatively, if anybody
Yes I have used it and it looks like this:
exten = s,1,Dial(Local/[EMAIL PROTECTED])
exten = s,2,Goto(s-${DIALSTATUS},1)
On 4/2/06, Stuart Elvish - Dallas Delta Corporation Pty Ltd
[EMAIL PROTECTED] wrote:
Hi,
I have searched the web and found some basic information about the
LOCAL/ channel.
Actually you are wrong about $$$ Dovid, I did not charge one penny for it.
On 4/2/06, Dovid Bender [EMAIL PROTECTED] wrote:
My guess (it's only a hunch) as to why he asked for
off list is for $$$. Usually when you see that the
person wants $$$. I know there is a way of doing it in
asterisk.
There is a reason why I am posting it off list and not because of money.
On 4/2/06, Miles Scruggs [EMAIL PROTECTED] wrote:
I would also like to know how to do this, it really defeats the whole
purpose of the list if you reply off list.
Please post that to the list.
Miles
Shaun wrote:
um ok, well do you mind posting it off list to myself, if you haven't
caught it I am interested.
Miles
Tom Vile wrote:
There is a reason why I am posting it off list and not because of money.
On 4/2/06, Miles Scruggs [EMAIL PROTECTED] wrote:
What I do is the following and keep in mind I only use one register
statement with my provider:
exten = 18665551234,1,SetVar(FROM_DID=18665551234) ;
exten = 18665551234,2,Goto(from-pstn,s,1) ;
exten = 5185551234,1,SetVar(FROM_DID=5185551234) ;
exten =
um ok, maybe not since you seemed a bit rude.
On 4/2/06, Miles Scruggs [EMAIL PROTECTED] wrote:
um ok, well do you mind posting it off list to myself, if you haven't
caught it I am interested.
Miles
Tom Vile wrote:
There is a reason why I am posting it off list and not because of money.
Nicolas,
Do you have any idea what this will cost and when it might be released?
Thanks,
Doug
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nicolás Gudiño
Sent: Friday, March 31, 2006 7:50 PM
To: Asterisk Users Mailing List -
I would like to know which extension number is engaged in a call.
show channels shows me:
*CLI show channels
Channel Location State
Application(Data)
SIP/asterisk.elmit.com-0 [EMAIL PROTECTED]:2Up
Echo()
SIP/8807-066
eh? what have I done that is rude, I never even made comments about money?
Tom Vile wrote:
um ok, maybe not since you seemed a bit rude.
On 4/2/06, Miles Scruggs [EMAIL PROTECTED] wrote:
um ok, well do you mind posting it off list to myself, if you haven't
caught it I am interested.
Miles
OK, enough of this.. No reason to bicker about something like this.
Here is the URL.
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+cmd+Dialdiff=57
For those of you that do not have a working web browser or cand find it with
Google here is the text.
Dial macros
Hello List!
I wanted to share to everyone the following compatible
connectivity products that my company installed in our
Asterisk based soft switch. I already sent these to
the Asterisk.org site many days ago but for some
reason they still have to post it.
1. Sangoma A101 single port E1/T1/PRI
Hi,
Is there a way to do a ZapBarge, but where the person doing the
barge-in would be able to talk to the agent only (whispering)?
Thanks,
Andre Courchesne - Consultant
http://www.net-forces.com
___
--Bandwidth and Colocation provided by
Neofita wrote:
-- channelsOpen = 1
There is only ONE channel open. This should be a huge alarm to you.
Jeremy McNamara
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options
The 'sip show channels' and 'show channels' command aren't exactly easy to
interpret, especially if one of the numbers has pic codes and rate centers
inserted (the rest is truncated on the output), or you have a proxy involved in
the call. Wish someone with some C knowledge would fix that.
[EMAIL PROTECTED] wrote:
I am not sure if this debug message is enough information.
Try to do what I told. Switch to another H323 channel driver and see
what happens. Try first chan_oh323.
So instead of solving his configuration problem he should try a new
channel driver?
Michael
Great thanks
Steven Job wrote:
OK, enough of this.. No reason to bicker about something like
this.
Here is the URL.
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+cmd+Dialdiff=57
For those of you that do not have a working web browser or cand find
it with Google
I have some troubles with ASTCC. TO often the in-use flag remains set.
I would like to find a solution, where astcc.agi checks automatically if
THIS user is in a call rather than to check the flag.
If that is not possible, I would like to have an extension to dial to,
and it will after
72 matches
Mail list logo
