I have [EMAIL PROTECTED] 2.5 running in my network. VEGA 50 Analog is used
as the gateway between PSTN VoIP. I am able to dial out to PSTN with out
any issues. When a PSTN caller calls to my VoIP network he is able to reach the
extensions. But even after the caller (PSTN) disconnects, the
Anyone know why I would be getting this error? All calls go
through without problem but I get the following message:
fixlocalprefix: Could not parse
/etc/asterisk/localprefixes.conf
___
--Bandwidth and Colocation sponsored by
From the command prompt type: help-aah
This will give you a list of commands to
change passwords. For example:
Commands Descriptions
---
config set the local time
zone and keyboard type
netconfig configure
I can't comment on anything but AMP. I've found it very easy to use.
Most recently, I've switched to [EMAIL PROTECTED] 1.5 which includes AMP +
other tools and found this package to be very interesting and intuitive.
I believe an SMB could manage Asterisk with a little guidance using the
[EMAIL
Can someone tell me how to do this...Given the following line:
exten = *97,3,VoicemailMain([EMAIL PROTECTED])
Is it possible to add some logic to manipulate the CALLERIDNUM to send
back 801 even if the extension is 601 and 901 even if the extension is
701? I have 2 branch offices where users
regards,
srsergio
-Mensaje original-
De: Chad Brown [mailto:[EMAIL PROTECTED]
Enviado el: miércoles, 31 de agosto de 2005 8:48
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] Manipulate CALLERIDNUM
Can someone tell me how to do this...Given
Im publishing tftp through my firewall to support
external Cisco 7960 sip phones. I know that the primary port is 69 for tftp.
However, tftp also uses secondary ports ranging from 1,0XX to 30,XXX. ( A broad
range) In an effort to limit the secondary ports that are opened, some Windows
If you have an account you can try: http://control.sixtel.net This works
and they seem to be adding some features. My service still works.
However sixtel has been unable to tell me how much $ is available for
use. I'm not too confident at this point.
-Original Message-
From: [EMAIL
: 650.246.8900
F: 650.246.8901
E: carlos at race.com
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown
Sent: Tuesday, August 02, 2005
10:46 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TFTP
Secondary Ports
Im publishing tftp through my firewall
There are many analogue gateways to choose from:
http://www.voip-info.org/wiki-VoIP+Gateways
Does anyone have experience with several that could point me in the
right direction? I need 5-8 ports. At some point I see us going digital
but I'm not sure when TCO will make sense.
Advice based on real
I recently upgraded to 7.4 and the time setting continued to work. You
say you upgraded and still have the exact same SIPDefault.cnf and
SIPMAC.cnf that worked in 7.3?
Chad Brown - IdentityMine
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Lee
solutions need the Remote
SIP Connectivity module.
Chad Brown - IdentityMine
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, March 26, 2005 12:25 AM
To: 'Wilson Pickett'; 'Asterisk Users Mailing List - Non-Commercial Discussion
and
choppy sound. Since adding QoS and changing to the G729a codec 5 days
ago we haven't had an incident of either yet. Time will tell...
Chad Brown - IdentityMine
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris
Edwards
Sent: Wednesday, March 23, 2005 9:42
I am noticing that calls that come from our
IAX pstn gateway provider and terminate to our Asterisk IVR do not receive
ringing when an extension is dialed. For example:
An
inbound PSTN caller calls our number
Asterisk
answers and provides greeting
PSTN
user dials
-Original Message-
From: Chad Brown
Sent: Tuesday, December 21, 2004 8:02 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dialplan help - Can dial any user but not
thePSTN
Flynn,
Yes, that makes sense. However, in my case I have incoming calls
arriving on an IAX channel from
Yes...Crystal.
Thanks Flynn
-Original Message-
From: el Flynn [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 21, 2004 10:31 PM
To: Chad Brown
Subject: Re: [Asterisk-Users] Dialplan help - Can dial any user but not
thePSTN
Chad Brown wrote:
Flynn,
You are being patient with me
without the duplication. Thoughts?
Chad Brown -
IdentityMine
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, including all
parameters. The UAC MUST then place the remote target URI into the
Route header field as the last value..
200OK Sent to the Asterisk by the SIParator. (Bad Call)
SIP/2.0 200 OK
To: sip:[EMAIL PROTECTED];tag=3307485377-144837
From: Chad Brown sip:[EMAIL PROTECTED
-Commercial Discussion'
Subject: RE: [Asterisk-Users] chan_sip changes affecting ACK? - Bug?
Have you opened a bug up?
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chad Brown
Sent: Monday, October 25, 2004 12:02 PM
To: Asterisk Users
Answering with capability 0x2(GSM)
Answering with capability 0x8(ALAW)
Answering with non-codec capability 0x1(G723)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK6016290f
From: Chad Brown sip:[EMAIL PROTECTED];tag=as2041d236
:
Info:sipfw:recv from
10.10.0.6: ACK sip:[EMAIL PROTECTED]
SIP/2.0
Asterisk sip debug says:
Transmitting:
ACK
sip:[EMAIL PROTECTED]
SIP/2.0
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK4441b7b5
Route: sip:[EMAIL PROTECTED]
From: Chad Brown
sip:[EMAIL PROTECTED];tag=as490d60cd
To: sip:[EMAIL PROTECTED
For some reason native bridging is not working since I upgraded to
version 1.0. Any ideas why?
-- SIP/10.10.0.5-0719 answered SIP/101-81c1
-- Attempting native bridge of SIP/101-81c1 and SIP/10.10.0.5-0719
Device SIP/10.10.0.5-0719 simply drops the call after only a couple
I have a customer that wants to try the
exact same thing next month. Unfortunately I dont have any advice for
you at this time. However, if the PIX doesnt end up working for you I
can tell you that Ive had excellent success with the INGATE product
line. (Both Firewall and Firewall
Chuck,
The first thing I would do is to upgrade the load to version 6 or
higher. I'm running the latest...version 7.2. (I'm very happy with it)
Are you using TFTP to load the configuration or manually configuring the
7960? I know it's a pain to setup TFTP just for a quick test. However,
it's
Yes, that's tough. A couple things though...
1. To be fair...My 3.2 load did work against Asterisk. I just feel that
troubleshooting should begin with the latest bug fixes applied if
possible.
2. You may be able to contact Cisco technical support to get the latest
firmware / files. Before I put
Is anyone working on a Fedora Core 2 RPM?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Boehnlein
Sent: Thursday, September 23, 2004 12:57 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
Hello,
Straight
.
Thanks,
Chad Brown - IdentityMine
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Shawn Dillon
Sent: Tuesday, September 21, 2004
2:39 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk
, ISA Firewall/VPN , STUN and other issues
I have just finished compiling
- Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Timing source on SMP system
try zaprtc from www.junghanns.net. Works fine in my SMP systems
- Original Message -
From: Chad Brown [EMAIL PROTECTED]
Date: Sat, 18 Sep 2004 20:23:54 -0700
Subject: [Asterisk-Users] Timing source on SMP
try it until monday.
Matthew
- Original Message -
From: Chad Brown [EMAIL PROTECTED]
To: Michael Bielicki [EMAIL PROTECTED]; Asterisk Users Mailing
List -
Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Sunday, September 19, 2004 1:13 PM
Subject: RE: [Asterisk-Users] Timing source on SMP
,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Sunday, September 19, 2004 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Timing source on SMP system - Disable RTC
forzaprtc
Chad Brown
] On Behalf Of Kristian
Kielhofner
Sent: Sunday, September 19, 2004 6:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Timing source on SMP system - DisableRTC
forzaprtc
Chad Brown wrote:
Kristian,
I have 2 X100P cards but neither work on my Compaq
I need a timing device for the DL360G2 for conferencing and
meetme. For a timing device I have 2 X100P cards but neither will work in my
DL360G2. The system will not even boot with either card in the system. Other
PCI cards seems to work fine. I called Digium support and was told that
-Users] Timing source on SMP system
try zaprtc from www.junghanns.net. Works fine in my SMP systems
- Original Message -
From: Chad Brown [EMAIL PROTECTED]
Date: Sat, 18 Sep 2004 20:23:54 -0700
Subject: [Asterisk-Users] Timing source on SMP system
To: Asterisk Users Mailing List - Non
..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown
Sent: Thursday, September 16, 2004 1:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; William
Suffill
Subject: RE: [Asterisk-Users] ztdummy on Fedora Core 2
Standard
-Original Message
- Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sip Outbound Proxy
Chad Brown wrote:
How do you configure an outbound proxy for Asterisk? If the sip call
is
not local I want everything to go to a designated sip proxy.
In the standard chan_sip, there's no support for outbound proxy
I followed the Wiki instructions to get zaptel to work on
Fedora core 2. It looked like everything went perfect including the loading of
ztdummy. However, I am having meetme and MOH problems synonymous with ztdummy
not loading. Take a look at my lsmodAny ideas? (I am running stable
] On Behalf Of Chad Brown
Sent: Wednesday, September 15,
2004 12:33 AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] ztdummy
on Fedora Core 2
I followed the Wiki instructions to get zaptel to work on
Fedora core 2. It looked like everything went perfect including the loading of ztdummy.
However, I am
Subject: Re: [Asterisk-Users] ztdummy on Fedora Core 2
ztdummy should be able to work natively on the 2.6 kernel w/o need of
usb. I
use it on a fc2 box single processor w/ a 2.6 kerenl w/o issue and a rh
9 w/
2.4 w/ usb
- Original Message -
From: Chad Brown [EMAIL PROTECTED]
Date: Wed, 15 Sep
How do you configure an outbound proxy for Asterisk? If the
sip call is not local I want everything to go to a designated sip proxy.
Thanks,
Chad
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
It looks like chan_sip2 may solve my problem with
outboundproxy support. However, I am having problems getting the solution
installed. From what I understand these are the tasks
Add
chan_sip2 to the channels/Makefile
* Rename the file downloaded to chan_sip2.c
* make / make install
*
Any advice on a good conference phone that works with
Asterisk? I like the Cisco line and was wondering if anyone has used the 7935
or 7936 phones. From what I can tell they dont have a sip load. Has
anyone verified this or gotten an ETA from Cisco?
Chad
I've been in discussions with broadvoxdirect.
[EMAIL PROTECTED] has been encouraging our company to sign up. I have
yet to do so.
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, August 28, 2004 3:43 PM
To: Asterisk
Jerry,
If your talking sip phones...
I am using the Cisco 7960 phones and love them. The quality and
stability against Asterisk have been excellent.
Chad Brown - IdentityMine
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Roy
Sent: Friday
Hi James,
I had the same problem. Setting echotraning=yes helped me.
Also, like many people I had to tweak the rxgain and txgain. For me
these setting worked well.
rxgain=0.0
txgain=-4.0
If I tried to increase the rxgain SIP echo got really bad. Although I
could hear the PSTN caller much
Does anyone know what ports need to be opened up when using TFTP to
configure Cisco phones?
The configuration below doesn't seem to work. However if I open
everything up the files comes down just fine.
iptables -A INPUT -p udp -s 0/0 --dport 69 -j ACCEPT
Thanks for your help!
Chad
Notice Zap/g2
-- Executing Dial(SIP/chad.brown-d1ac, Zap/g2/9528737) in new stack
Aug 1 00:42:43 NOTICE[1200884528]: app_dial.c:714 dial_exec: Unable to
create channel of type 'Zap'
== Everyone is busy/congested at this time
Does anyone know why Asterisk wants to use group 2 regardless of how I
I was just in Atlanta and stayed at the Marriott Marquis. It was a nice
hotel but it was not cheep. I spent 200 for a last minute booking.
However, there is a train that runs from the airport and terminates a
block away. (5$ one way)
The Hilton and Hyatt are next door.
Chad
-Original
I have 2 SIP phones (Cisco 7960 XTen) behind a NAT
provided by a Linksys firewall that supports UPnP. The Asterisk server
has a public IP. Here are the problems that I am having with this configuration
The 2 SIP phones can call
MeetMe and have a conference but cannot call each
.
www.servers-r-us.com
(775) 284-5800 ext 115
On Mon, 24 May 2004, John Fraizer wrote:
Chad Brown wrote:
I have 2 SIP phones (Cisco 7960 XTen) behind a
NAT provided by a
Linksys firewall that supports UPnP. The
Asterisk server has a public
IP. Here are the problems that I am
Does anyone have experience setting up * to accept anonymous
sip UAs and the dumping the call into IVR? Im thinking this would be a
good way to have customers call us without creating an extension. So for my tests
have been focused on providing internal functionality.
Thanks,
Chad
We manage our own VOIP solution using Asterisk.
Has anyone had success with an IP-PSTN provider? I'm looking for someone to terminate
SIP calls to the PSTN in the Seattle, Washington area. (vice-versa as well if possible)
Yes, I could do it myself via asterisk and digium cards but I would
Is something different about the latest implementation of voicemail in
asterisk?
I have been away from asterisk for about 6 months and finally have a
chance to take a second look at it. From what I remember all you needed
to do to setup voicemail was to perform som basic modifications to the
Thanks! Yes this is what I was looking for. Since I have never applied
an Asterisk patch or Linux patch for that matter I am very interested in
the instructions for installing the patch and prompts.
The links suggests that instructions are included in the .tar but I did
not could not find them.
We have 2 offices. One is in the US and the other in India. We are
testing Asterisk as a possible solution. Does anyone have advice for the
preferred architecture when dealing with the latency from the US to
India?
For example, although it is not a requirement to have an Asterisk server
in India
I am new to Asterisk and looking as a solution to our
offshore development team. We are currently testing SIP solutions and having a
difficult time maintaining even mediocre care quality. It seems that we are
much better off just using Yahoo Conferencing than implementing our own
solution.
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