Thanks for the informations.
Regards,
Freddy
Go to http://www.voxzone.com/osCommerce
This is from Singapore.
Hi,
Does anyone know where I can get the X100P card and VoIP phones
in
Singapore? I really want to buy it.
___
Hi all,
As we know, we can use dial plan to control the flow of the call
making process. However, I want to know how to control during the
registration process. Say, in ARA, during the registration process,
there is update sql statement in the user table. And I want to store
more information
5 jul 2006 kl. 13.46 skrev Roger Schreiter:
Hi,
I did not yet study the newest chan_sip.c versions, but
it seems, that chan_sip treats mysql-peers different from
other peers, concerning the variable canreinvite.
If this variable is not explicitely set for a peer or user in
sip.conf, the
6 jul 2006 kl. 04.09 skrev Sharon Lim:
hi,
Is it possible to have same sip context but refering to different
username and context?
Is it a must to have username(test) and the sip context [test] the
same?
what the different with username sip context?
can username variable be
6 jul 2006 kl. 09.01 skrev unplug:
Hi all,
As we know, we can use dial plan to control the flow of the call
making process. However, I want to know how to control during the
registration process. Say, in ARA, during the registration process,
there is update sql statement in the user table.
Turn up - it's always good
Pint on Punt (Punt road)
Tonight's lecture - QOS.
later,
Paul Hales
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On Jul 5, 2006, at 7:03 PM, Josh Reineke wrote:
I work for a medium size business in Anchorage, AK running two
installations with about 30 handsets a piece. They've both been in
service for a couple of years.
I'm in Seattle fairly frequently, being it's the metropolis closest to
Anchorage.
Thanks Peter,
Peter Bowyer wrote:
On 05/07/06, Kai Fürstenberg [EMAIL PROTECTED] wrote:
Just dial the international number completely (e.g. for Germany 0049etc.)
In your extension above a number beginning with 011 is being dialed.
That is not an international number.
Where were you assuming
Patrick wrote:
snip
AMD's Opteron has more FPU power than Intel. FPU power is needed when
you do a lot of transcoding. So there is a difference besides AMD
being
cheaper.
Regards,
Patrick
On Jul 5, 2006, at 6:32 PM, varun wrote:
Thanks Patrick,
That is useful info for future.
Actually it
Has anybody managed to get the ACD function to work with Polycom IP501
phones and Asterisk?
I have used the trunk versions of libpri and zaptel and the
polycom_acd_functions -r30432 branch of asterisk, but it still will not
work.
Are there any special settings that need to be in the polycom
Hello,
I creat two context in extensions.conf:
[creat-in]
exten = 400,Dial(IAX2/400,20,tr)
exten = 400,2,Voicemail(u400)
exten = 400,10,Hangup
exten = 401,Dial(IAX2/401,20,tr)
exten = 401,2,Voicemail(u401)
exten = 401,10,Hangup
[steganux-in]
exten = 300,Dial(IAX2/300,20,tr)
exten =
Do you mean source code modification is the only way to do it?
In register message, I get the following information.
Authorization: Digest username=8766760539, realm=asterisk,
nonce=51cff2e4, uri=sip:203.193.26.234,
response=e6899541b2793d3a4a700ed220143322.
Content-Length: 0.
I want to get the
Hi all
At my home i have one rockwell modem
with chipset Rockwell 97 and i have one Linksys PAP2
I want to make small PBX at home for testing
So i want to use this modem for incoming and transfer call to PAP2 ports
auto attendent
can some one tell me is this modem works for incoming calls for
Hi,
Olivier Saulnier wrote:
Hello,
I creat two context in extensions.conf:
[creat-in]
exten = 400,Dial(IAX2/400,20,tr)
exten = 400,2,Voicemail(u400)
exten = 400,10,Hangup
exten = 401,Dial(IAX2/401,20,tr)
exten = 401,2,Voicemail(u401)
exten = 401,10,Hangup
[steganux-in]
exten =
Hello,
I have a requirement of bridging 2 sip connections via
asterisk, which has to be web based.
A person has to go to a webpage and enter his from sip
uri(say sip1) and enter another sip uri(say sip2). Upon pressing the connect
button, the webpage needs to send say a dial sip1
My concern on sip.conf , is user able to create their own extension . Therefore how to control the sip user account. My understand are, 1. You need to create either zap/sip/iax channel ~user account as a user.
2. Then to activate the account you need to have a extension number to ring the user
Hello
You can just use the Asterisk Manager API.
Its relatively easy to create this kind of application, just look at the
Originate function of the API.
http://www.voip-info.org/wiki/view/Asterisk+manager+API
Theres lots of examples for many
different programming languages.
Jon
ram wrote:
Hi all
At my home i have one rockwell modem
with chipset Rockwell 97 and i have one Linksys PAP2
I want to make small PBX at home for testing
So i want to use this modem for incoming and transfer call to PAP2 ports
auto attendent
can some one tell me is this modem
Tzafrir Cohen,
I'm not a linux expert, i just wanted to share what happened to me...
Only after new detection of Sound Board, i got the audio from Asterisk
services like MOH, Voicemail busy message ... working fine.
Probably there's a better faster way to run kudzu, i tried from
command line
Dinesh wrote:
Hello,
I have a requirement of bridging 2 sip connections via asterisk, which
has to be web based.
A person has to go to a webpage and enter his from sip uri(say sip1) and
enter another sip uri(say sip2). Upon pressing the connect button, the
webpage needs
Also have a look at .call
files.
You web app can just create a .call file and then move it to the right
location and asterisk will place the call
No manager interface needed.
Marnus van Niekerk
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
motorola ??
what is the model ?
iam looking PCI Cards
can some one suggest me,
iam Located in india, i see X100P cards charging more than 80$ here
so want to try with PCI modems and see if i can success
ram
On 7/6/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
ram wrote: Hi all At my home i have
Hi all!
How do I make Asterisk recognize fax calls and disconnect them?
Regards,
Evert
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I am building a server that accepts audio over SIP and returns
pre-recorded audio files based on the result of a recognizer. However,
I am not sure exactly how to establish and collect data over and SIP
connection. I literally only need a program (socket?) to establish the
connection and
Hi Friends,I am using Polycom IP 501 hard phone. Now, I want to confiugre my Polycom IP 501 phone with my Asterisk. I am unable to find any good link or tutorial for this. Please give a good link to configure my Polycom IP 501 phone with my Asterisk. Looking forward for your response. Thank
Asterisk comes with some demo config info in most .conf files. In SIP.conf
there is one written for Polycoms. Give it a try it works really well.
Joel
Asterisk IT
Hi Friends,
I am using Polycom IP 501 hard phone. Now, I want to confiugre my Polycom
IP 501 phone with my Asterisk. I am unable
On 7/6/06, Evert Meulie [EMAIL PROTECTED] wrote:
Hi all!
How do I make Asterisk recognize fax calls and disconnect them?
Regards,
Evert
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Ive looked for cheaper mobile voip pstn calls in Nigeria and the cheapest
ive found is about 18p/min or 28c/min. Im looking at providing for
cpmanies here cheaper calls comparable to call cards. Can someone give me
some ideas as to how i can do this. Can you negotiate prices with a
reliable
On Thu, Jul 06, 2006 at 11:57:15AM +0200, Evert Meulie wrote:
Hi all!
How do I make Asterisk recognize fax calls and disconnect them?
faxdetect=incoming in zapata.conf.
Set up an extension 'fax' that will simply Hangup().
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755
Hi,
I've forgot thos part, but default context contains:
[default]
include = demo
include = free-out
include = bri-out
include = steganux-in
include = creat-in
include = steganux-out
include = creat-out
include = commun ; contexte pour usage interne
I trie to creat a new context, nammed commun
Olivier Saulnier wrote:
Hello,
I creat two context in extensions.conf:
[creat-in]
exten = 400,Dial(IAX2/400,20,tr)
exten = 400,2,Voicemail(u400)
exten = 400,10,Hangup
This is incorrect. It should be:
[creat-in]
exten = 400,1,Dial(IAX2/400,20,tr)
exten = 400,2,Voicemail(u400)
exten =
Hi Olivier,
Olivier Saulnier wrote:
Hi,
I've forgot thos part, but default context contains:
[default]
include = demo
include = free-out
include = bri-out
include = steganux-in
include = creat-in
include = steganux-out
include = creat-out
include = commun ; contexte pour usage interne
I trie
Maxim Vexler wrote:
On 7/6/06, Evert Meulie [EMAIL PROTECTED] wrote:
Hi all!
How do I make Asterisk recognize fax calls and disconnect them?
Regards,
Evert
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Doug Lytle a écrit :
This is incorrect. It should be:
[creat-in]
exten = 400,1,Dial(IAX2/400,20,tr)
exten = 400,2,Voicemail(u400)
exten = 400,3,Hangup
Ok, i forgot priority!!
Best regards,
--
Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F:
Hie,
I trie to use a simply call forward, found on this mailing list (:-),
when i'm not near my phone:
i creat a global set:
olscell=123456789 ; my cell phone number
A macro for forwarding the call:
[macro-cell_user]
exten = s,1,Playback(Call_Transfer)
exten = s,2,Flash()
exten =
Hi,
recently im working on using meetme application in asterisk. i have
explored all the options for meetme application and i believe there is
no option for inviting a person to the conference while the conference
is on. is there any other way to do that?
Hi*,
I have the problem
that the cdr account sthe ringing seconds too.
normally it should
begin accounting when the asterisk gets a answer but it seems it is accounting
all the time since the sip connection from client to client is
established.
I use Oracle DB with
the cdr plugin from
Define invite.
Yelling across the office saying, Yo,
Dude! Dial ${CON}, works as an invite for me!.
If what you want is an automated invite
look at callout files and using creative dialplan options to Meetme App.
Snip
Hi,
recently im working on using meetme application in
Hello everyone,
I'm trying to set up an Asterisk machine with a quad-port BRI
Junghanns card, and I want to use the mISDN drivers.
I'm having some trouble configuring it: do I need to use CAPI drivers?
I haven't found good links, could you please provide some info?
Thanks in advance,
--
Andrea
Hello,
I look for documentation to use outboundproxy and
outboundproxyport in sip.conf.
I tried to add it in sip.conf but asterisk send
nothing to the proxy .
Asterisk lookup for an extension so 404 not found is
sent back.
I looked at
http://www.voip-info.org/wiki-Asterisk+config+sip.conf.
Tzafrir Cohen wrote:
On Thu, Jul 06, 2006 at 02:27:59AM +0100, Marco Mouta wrote:
I'm also surprised there is no faster way to run kudzu than to (yuck)
reboot.
How about just typing /usr/sbin/kudzu as root from a console?
W
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Hi,I installed G 729 and G 723 codecs and it works like magic. Download link http://kvin.lv/pub/Linux/Asterisk/built-for-asterisk-1.2-untested/
-KimOn 7/5/06, Levis Kimotho [EMAIL PROTECTED] wrote:
Hi,Ive installed freePBX and Asterisk and all is working well.
Ive created 2 extensions using
Hi,I installed G 729 and G 723 codecs and it works like magic. Download link
http://kvin.lv/pub/Linux/Asterisk/built-for-asterisk-1.2-untested/
-KimOn 7/4/06, Levis Kimotho [EMAIL PROTECTED] wrote:
Hi,Below is part of the log file
Jul 4 16:38:06 VERBOSE[5871] logger.c: dialparties.agi: Caller
how do i do that.let me tell u that im new to asterisk technology,
so ur gona have to walk me thru the solution. i do have a background in
programming so i can do whatever configuration for asterisk u want me
to do. tell me if there is any help on the internet about this. On 7/6/06, Alexander
After googleing extensively, I now have sip firmware (8.0.2SR1/8.0.3)
running on the 7941, 7961 and 7971 and I even have a SEP.cnf.xml that
seems to have everything and works (thanks to articles on
www.voip-info.org).
BUT the phones do not register correctly with asterisk.
Everything is
I would like to walk you through it but I
have much on my plate right now that requires my attention.
I will point you in the right direction.
Look at the menu options in MeetMe, the
exit menus in particular. These would allow you (admin) to press a key and exit
into the dialplan.
Per Møller wrote:
This line is missing on the 7941/7961/7971.
Anybody know why?
Without seeing any of your configuration files, I wouldn't be able to
guess. My 7940s and 7960s work just fine. Although, I'm running the
7.4 firmware.
Doug
-- Ben Franklin quote: Those who would give
I use the same firmware as you on your 7960's and SIP authenticates
ok. On our asterisk side we use the following config
[USERNAME]
type=friend
host=dynamic
canreinvite=no
username=USERNAME
secret=PASSWORD
callerid=Name of person switchboardnumber
The only gotcha i noticed at first was that on
Per Møller wrote:
This line is missing on the 7941/7961/7971.
Anybody know why?
Without seeing any of your configuration files, I wouldn't be able to
guess. My 7940s and 7960s work just fine. Although, I'm running the
7.4 firmware.
Doug
Hey Doug,
Yes my 7940 and 7960
Been testing a new spa941 with the latest firmware (sip to sip). I
noticed that unlike 7960's, if a user of a 7960 hangs up at the end of a
conversation, the 941 does not automatically hangup. Rather, the 941
sits there for about 5 seconds, then provides a fast busy. The 941 user
must
I'm in Williston, North Dakota and we have an office in Billings, MT. He's
right. We are 500 miles form civilization! :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday, July 05, 2006 10:00 PM
To: Asterisk Users Mailing
On Thu, Jul 06, 2006 at 09:52:57AM +0100, Marco Mouta wrote:
Tzafrir Cohen,
I'm not a linux expert, i just wanted to share what happened to me...
Only after new detection of Sound Board, i got the audio from Asterisk
services like MOH, Voicemail busy message ... working fine.
Probably
On Thu, Jul 06, 2006 at 03:19:13PM +0200, Andrea Spadaccini wrote:
Hello everyone,
I'm trying to set up an Asterisk machine with a quad-port BRI
Junghanns card, and I want to use the mISDN drivers.
That card is said to be exactly the same as the bero card. They both
look strikingly similar,
Per Møller wrote:
Hey Doug,
Yes my 7940 and 7960 using the 7.4 or 7.5 SIP firmware works fine and does
not use xml style config files.
I was looking for:
Asterisk side:
sip.conf
Cisco side:
SIPDefault.cnf
SIPMacaddress.cnf
Doug
-- Ben Franklin quote: Those who would
Hi,
i need to set the dtmf mode on my quintum tenor a400 gateway. i cant
dial any extension thru my normal digital phone which is connected to
asterisk thru the quintum gateway. it always falls to 's' extension. So
plz help
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Hi all !
i'm having a strange issue with an asterisk box behind a firewall
i'm trying to answer a sip call made to an asterisk box with a public ip
from another asterisk box behind a firewall
on the natted box i've put
externip=195.110.XXX.XXX
localnet=10.1.1.0/255.255.255.0
and on the phone
The 79X1 phones don't use the same configuration setups as the 79X0's.
They're the upgraded versions, using the SEPmac.cnf.xml files instead
of the SIPmac.cnf files.
On Thu, 2006-07-06 at 10:44 -0400, Doug Lytle wrote:
Per Møller wrote:
Hey Doug,
Yes my 7940 and 7960 using the 7.4 or
This was written for use with AAH but should work for you as well.
http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+7#722Polycom
On 7/6/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi Friends,
I am using Polycom IP 501 hard phone. Now, I want to confiugre my Polycom IP
501
On 7/6/06, Rich Adamson [EMAIL PROTECTED] wrote:
Been testing a new spa941 with the latest firmware (sip to sip). I
noticed that unlike 7960's, if a user of a 7960 hangs up at the end of a
conversation, the 941 does not automatically hangup. Rather, the 941
sits there for about 5 seconds, then
Doug,
Cheer up! There's some great beer brewed in Montana! Have a Moose Drool and get down to some creative resume re-inventing.
On 7/6/06, Kevin Savoy [EMAIL PROTECTED] wrote:
I'm in Williston, North Dakota and we have an office in Billings, MT. He'sright. We are 500 miles form civilization! :)
What is the current recommended version of firmware for SIP on
7960/7940's. I was reading through some of the stuff on voip-info and
it looks like the 8.x's have pretty serious bugs in regards ti *. Thanks.
PA
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On Thu, 2006-07-06 at 10:55 -0500, Peder @ NetworkOblivion wrote:
What is the current recommended version of firmware for SIP on
7960/7940's. I was reading through some of the stuff on voip-info and
it looks like the 8.x's have pretty serious bugs in regards ti *. Thanks.
PA
We stick
Rizwan Hisham wrote:
Hi,
i need to set the dtmf mode on my quintum tenor a400 gateway.
You might want to check the a400 manual on how to do that.
i cant dial
any extension thru my normal digital phone which is connected to asterisk
thru the quintum gateway. it always falls to 's' extension.
Peder @ NetworkOblivion wrote:
What is the current recommended version of firmware for SIP on
7960/7940's. I was reading through some of the stuff on voip-info and
it looks like the 8.x's have pretty serious bugs in regards ti *. Thanks.
Stay with the v7.x version until the newer stuff
Steve Davies wrote:
On 7/6/06, Rich Adamson [EMAIL PROTECTED] wrote:
Been testing a new spa941 with the latest firmware (sip to sip). I
noticed that unlike 7960's, if a user of a 7960 hangs up at the end of a
conversation, the 941 does not automatically hangup. Rather, the 941
sits there for
Or you could use web-meetme, it has this feature.
On 7/6/06, Alexander Lopez [EMAIL PROTECTED] wrote:
I would like to walk you through it but I have much on my plate right now that requires my attention.
I will point you in the right direction.
Look at the menu options in MeetMe, the exit
Yep, forgot bout that.
Or you could use
web-meetme, it has this feature.
On 7/6/06, Alexander Lopez [EMAIL PROTECTED] wrote:
Snip, snip.
Chop Chop.
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Hate to drag this one back up, butit's happening again.
Overview of architecture:
Dell poweredge 2850, running fedora core 4, asterisk 1.2.7.1, zaptel
1.2.5, and sangoma wanpipe 2.3.4 drivers. T1 interface card is the
sangoma a104d with onboard echo can.
Server is located in our data
Where may I find the 7.4 firmware for 7940? I was only able to find the 8.2
at cisco's website.
F
On 7/6/06 11:05 AM, Aaron Daniel [EMAIL PROTECTED] wrote:
On Thu, 2006-07-06 at 10:55 -0500, Peder @ NetworkOblivion wrote:
What is the current recommended version of firmware for SIP on
Does anyone know if Asterisk Home (2.7) runs OK on a 64 bit architecture?Thanks.
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Hello everyone,
I've set up an asterisk box with basic PBX features (DiD, MoH, MoT,
Blind and Attended Call Transfer, PickUp, ecc.) for 10 Cisco 79xx (7912
and 7960) with SIP image (8.0). PSTN gateway is done using a Cisco
AS5350 with two ISDN PRIs connected to Asterisk via SIP. Between the
On Jul 6, 2006, at 6:01 AM, Peder @ NetworkOblivion wrote:
Is there a buddies feature on the Cisco phones, like there is on
the Polycom? If not, how are people getting around the issue where
a receptionist wants to see who is on the phone? Or are they just
living with the limitation?
-Original Message-
Thanks for reading,
Wes
___
Please reply with the output of the following:
lspci -vv
lspci -vv | grep IRQ
lspci
cat /proc/interrupts
Thank you.
Andrew
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Don
Sent: Wednesday, July 05, 2006 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] intel vs amd motherboards
If you want to
The 8.2 firmware works just fine (asterisk 1.2.6). I have had exactly
zero problems with it, nothing even weird about it. Pretty trouble-free
IMO.
I believe the phone that doesn't work quite right with the 8.2 SIP image
is the 7970. I have probably 20 7940s/7960s, all running the 8.2 SIP
image
snip
Server load is averaging around 20%, plenty of memory, disk space, and
bandwidth available. No QOS running on network. ulaw is the primary
codec. Server is stable, and there are no extraneous services
running, save mysql and httpd. Even running a processor intensive
query doesn't trigger
Here ya go:
lspci -vv
---
00:00.0 Host bridge: Intel Corporation E7520 Memory Controller Hub (rev 09)
Subsystem: Dell: Unknown device 016d
Control: I/O- Mem+ BusMaster+
Your problem is intermittent. It is probably Network related as if you
reboot that problems may or may not comeback.
In addition to the lspci stuff requested. Have you checked your
fiberlink. Is it possible that something or someone is saturating the
link with Virus/Spy/PtP Ware???
SIP doesn't
Have completed a rough write up and I am having a linux/asterisk guru look it over. Once I submit it I would like some feedback on anystuff I left out or need to change.
Thanks
Julian
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Got Mediatrix
So you need a divide and conquer strategy here:
1. Is it Asterisk or the WAN? This should be easy enough to test for. Do
call dropouts happen in your datacentre? If not, your Asterisk install is
good. My money's on the 10mbit WAN pipe, and that's what I would be
focussing on.
2. If it's the WAN,
Pele Zico wrote:
Ive looked for cheaper mobile voip pstn calls in Nigeria and the cheapest
ive found is about 18p/min or 28c/min. Im looking at providing for
cpmanies here cheaper calls comparable to call cards. Can someone give me
some ideas as to how i can do this. Can you negotiate
The mail system somewhere seems to have eaten some of the digetst versions of this list that are sent to me (jumped from 24 to 29). So - in case this didn't make it out, just expressing my interest. Were there many others around here who responded that I must have missed?philippePhilippe
Thanks for the quick responses everyone.
To answer some of the questions posed:
The main traffic going over this pipe is voice, with a small amount of
web traffic as well. There are 60 total users, 5 of which access
anything other than what is on their LAN up there. In any case, we
are not
Hello List,
I work for VoiceIP Solutions in Seattle(Asterisk Provider) and I am willing
to help set the group up also. Please email me and we should get a core
group together and figure out how we want to handle it.
Thanks,
-Lauren M.
- Original Message -
From: Josh Reineke
Also, when I connect to the server locally (the server is in the room
next to me, in other words, and i have 1 Gbit of bandwidth all the way
to the back of the server, I still get call dropouts.
However, this IS the only server (of 8 total, all in the same rack and
connected to the telco via
Hi,
I'm getting a lot of these messages in my asterisk 1.2.9. The clients
are all Xlite Xten softphones. Isn't this message supposed to be part
of hardphones?
Got SUBSCRIBE for extensions without hint. Please add hint to 1001 in
context xlitephones
extensions.conf
exten =
Can I please have some orientation as to how to plot in a graph the
data presented by fxotune -d parameter?
Thanks,
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Colin,
Very good points, and you are right, I need to start tracking what has
been done.
A bit of history - this server was very unstable when running Digium
hardware - every day or two, it would kernel panic and lock up,
requiring a manual reboot. The other servers had issues as well, and
ALL
I have recently build 2 machines, one with an Intel Pentium Dual Core
CPU, and one SATA HDD, and the other with a single AMD 64 bit CPU, and
a RAID 1 w 2 SATA HDD. Both costed the same even though the AMD had 2
HDDs. Here are the show translations from both:
Intel Dual Core machine:
pbx*CLI show
in the rtp.conf you specify range of ports, this range should be
forwarded on the firewall to the asterisk box.
I have asterisk running on 192.168.0.250 ip and connect to broadvoice
server. here is my iptables rule:
-A PREROUTING -i eth0 -s 147.135.0.128 -p udp -m udp --dport 10010:10013
-j DNAT
What brand/model phones are you using.
--
~Shaun
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Polycom 301/501/601, and SPA9xx
On 7/6/06, Shaun [EMAIL PROTECTED] wrote:
What brand/model phones are you using.
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~Shaun
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On Thu, 2006-07-06 at 12:44 -0700, Shaun wrote:
What brand/model phones are you using.
Aastra all models
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Dave Cotton [EMAIL PROTECTED]
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Im
having an issue where Asterisk hangs up a call (either party hangs up) but the
telco side of the T1, both the local company and ATT, does not receive the
hangup signal from Asterisk. Therefore Asterisk thinks the channel is available
but its still off-hook on the telco side. I have
Polycom 501
Grandstreams are junk. (I have had bad experiences with them)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shaun
Sent: Thursday, July 06, 2006 12:45 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] for you guys setting up
calvis wrote:
Grandstreams are junk. (I have had bad experiences with them)
The former doesn't necessarily derive from the latter :-)
Others of us have found them to be an excellent low-cost solution that
puts VoIP in places it otherwise would not be economical to deploy.
Everyone's
Polycom 601. The screen on the 501 is too small. You can't even fix a full 7
digit number on the screen. What was Polycom thinking when they did that? They
also don't have the microbrowser.
-Original Message-
From: calvis [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 06, 2006 2:01
Kevin Savoy wrote:
I’m having an issue where Asterisk hangs up a call (either party hangs
up) but the telco side of the T1, both the local company and ATT,
does not receive the hangup signal from Asterisk. Therefore Asterisk
thinks the channel is available but it’s still off-hook on the
On 7/6/06, Kevin Savoy [EMAIL PROTECTED] wrote:
I'm having an issue where Asterisk hangs up a call (either party hangs up) but
the telco side
of the T1, both the local company and ATT, does not receive the hangup signal
from
Asterisk. Therefore Asterisk thinks the channel is available but it's
Good point. I am using Asterisk 1.2.9.1, Zaptel 1.2.6. libpri 1.2.3. I had
the same issue with Asterisk 1.2.7.1, Zaptel 1.2.5 and libpri 1.2.2.
My Zapata.conf looks like this:
[channels]
context=default
musiconhold=default
resetinterval=60
;ATT T1's
group=1
switchtype=national
signalling=em_w
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