[EMAIL PROTECTED] wrote:
BUT, I am now able to call both phones at the same time and they
both ring. This sounds like Asterisk is blocking multiple
connections from the same IP Address? How do i go about fixing
this?
Your NAT bindings are probably expiring. You need to have your phones
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
BUT, I am now able to call both phones at the same time and they
both ring. This sounds like Asterisk is blocking multiple
connections from the same IP Address? How do i go about fixing
this?
Your NAT bindings are probably expiring. You
[EMAIL PROTECTED] wrote:
FWIW, I've seen several cases similar to this (not with asterisk
though) where the nat box itself had a limit (bug or otherwise,
unknown) as to how many devices behind the box could use a specific
port number.
Ahhh, thats right! We have seen this many times
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
FWIW, I've seen several cases similar to this (not with asterisk
though) where the nat box itself had a limit (bug or otherwise,
unknown) as to how many devices behind the box could use a specific
port number.
Ahhh, thats right! We have
[EMAIL PROTECTED] wrote:
Do you have these values set?
externip
localnet
localmask
My Asterisk Is actually on a public IP Address, but I do have externip
configured
externip = 64.72.x.x
;localnet = 0.0.0.0
;localmask = 0.0.0.0
-Original Message-
From: Harold Workman
digit. And the 28155512T is actually matching 8
digits which means if a call came in as 2815551234 it would match more
digits to dial peer voice 3.
---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
[EMAIL PROTECTED] wrote:
The call
Does asterisk block multiple register attempts from the same ip address?
Heres my example
64.72.100.139 = Asterisk
166.67.78.9 = NAT(PAT) IP Address on Cisco Pix Firewall
pbx*CLIsip show peers
2815551212/2815 166.67.78.9 D N 255.255.255.255 63289
Unmonitored
2815553434/2815
, no connection to PSTN or
ISDN - just receiving 'calls' via SIP, answering them and playing a
message.. Can anyone point me to a walkthrough that is focused on SIP
alone? Thanks.[Harold Workman]
http://voip-info.org/wiki-Asterisk+config+sip.conf
Hi,
I am having a problem when I add multiple phones behind a Symmetric
Firewall. Heres my situation.
11am - Phone A registers with *
11:01am - test call to Phone A. Call works fine.
11:02am - Phone B registers with *
11:03am - test call to Phone A fails, test call to phone B works fine.
Sorry for the second email, but be a little more clear. I am making the
test call to Phone A and Phone B from PSTN Cisco Gateway. Calls are routed
to * via SIP from Cisco 7206
PSTN --- Cisco --- * --- IP Phone
Harold Workman wrote:
Hi,
I am having a problem when I add multiple phones
[EMAIL PROTECTED] wrote:
hi;
can anyone provide me with some pointers on how to integrate ser with
asterisk.
http://www.voip-info.org/wiki-Asterisk+at+large is not very useful
especailly on the setup of ser.
thanks
ms
___
Asterisk-Users
When was this implimented? I am currently running
CVS-HEAD-07/07/04-09:34:29
---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of brian
Sent
source pbx
that are not familiar with Asterisk. There can never be too much
documentation. What are the signate's thoughts on having a lower cost PDF
file for download?
---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
-Original
NO NO NO.change your EMAIL ADDRESS. You cannot have a . in your email
address. Try [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
This is all I get. Can I send /msg -messages?
My name has no symbols like ¤%¤%¤%...
* Connecting to irc.freenode.net (6667)
-
-irc.freenode.net- ***
[EMAIL PROTECTED] wrote:
When was this implimented? I am currently running
CVS-HEAD-07/07/04-09:34:29
---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
[EMAIL PROTECTED] wrote:
Looking at the wiki on QoS is very helpful.
But just wondered if there is anyone who
would want to comment on ip rtp header-compression?
Is it a good thing, does it help on T1 or slower connections?
Also why does Cisco often suggest the MultiLink thing?
Heres the
what does that have to do with an overpriced book?
and i agree with Joe. With this book sourcing most of the documentation
directly from wiki, why pay for something thats free? Id rather donate $49
to keeping wiki free to the enviroment.
---
Harold Workman
CCNA, CCNP
Cytel
exten = s,9,ResponseTimeout,10
exten = s,10,Background(/usr/src/asterisk-sounds/sounds/pls-stay-on-line)
exten = s,11,Monitor(wav,,m)
exten = s,12,Queue(cytelcs)
---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
try cvs checkout -D "04/01/04
asterisk-addonswith the latest cvs of
asterisk
harold
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of rich
allenSent: Tuesday, July 06, 2004 11:22 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] compiling
Ive seen this question asked a few times but with no resolving answer. Im
running CVS-06/24/04-22:20:31, on RedHat Fedora 1, and cannot get chan_cssp
to compile. Im getting
Now compiling chan_sccp.c 695 lines
chan_sccp.c:50: warning: type defaults to `int' in declaration of
and for some reason he never answers the phone or returns calls. Its been
4 days that ive been trying to buy some of their phones, with no luck.Id
purchase them online, but not at their list price when their resellers give
10% discounts.
---
Harold Workman
CCNA, CCNP
Cytel
and i see your in the same region as myself in houstongood luck =)
let me know if you have success with them, for im still interested in their
quality of phones.
---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
try
cvs checkout -D 04/01/04 asterisk-addons
with the latest cvs of asterisk
harold
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Darrin
Johnson
Sent: Wednesday, June 30, 2004 2:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
as well. Are you conecting to the 7200 via serial or ip? sip?
---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com
be
0x2102. And again, i would look for tracebacks...it could either be a
memory issue or a bug in the IOS. But you will know if you get console
access to the router as u bring up the asterisk...
---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449
of IOS that is
optimized for VOIP. I ran 12.2(x) and could never get DTMF tones to work...I
upgraded to 12.3(x) and fixed the DTMF tones but now all FOIP calls fail.
--
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
Quoting Brian Wilkins [EMAIL
Tommy,
Thanks, how do i get the older version of asterisk-addons?
--
Harold Workman
Quoting T. Chan [EMAIL PROTECTED]:
Hi,
I got the same thing, so what I did was for the asterisk-addons, I used CVS
April instead of the most current CVS and it worked. Of course, I would have
liked
Tommy,
I reverted asterisk-addons to 04/01/2004 and I was able to compile it with the
latest asterisk CVS. Your a lifesaver. Ive been pondering over this problem
for over a week now. Thanks!
--
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
-server and devel upgraded at version 4.0.20 on a Fedora Core 1. I
would really love to have mysql support
Harold Workman
This message was sent using IMP, the Internet Messaging Program
[EMAIL PROTECTED] wrote:
Hello,
You need to have your versions of asterisk and asterisk-addons
match, if you are getting a lock error. Download both from the cvs
and compile. This should fix your error.
- Original Message -
From: Hekuran Doli [EMAIL PROTECTED]
To: [EMAIL
[EMAIL PROTECTED] wrote:
Here is the jist: Freeradius is up running and
functional using SIP Express radius how to. My
asterisk box has app radius installed. Is there
any documents on how-to link asterisk to freeradius?
documentation is lacking on app radius, at least
not as detailed as I
Hi,
I have setup agent 1000 in agents.conf, and the queue cytelcs in
queues.conf, and my extensions.conf file looks like
exten = 28,1,AgentCallbackLogin
exten = 9665,1,AddQueueMember(cytelcs|SIP/2815691212)
exten = 9665,2,Playback(agent-loginok)
exten = 9665,3,Hangup
I am able to log into the
Message-
From: Harold Workman [mailto:[EMAIL PROTECTED]
Sent: 22 June 2004 18:54
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AgentCallbackLogin - invalid extension
Hi,
I have setup agent 1000 in agents.conf, and the queue cytelcs in
queues.conf, and my extensions.conf file looks like
Just a quick question. I setup Directory dial by name, and I read it looks
at the Voicemail config to determine who you want to connect to. The thing
I dont like is when it finds a match it reads the extension instead of their
name. Is there a way to have it read the name in the voicemail
Bruce Komito wrote:
Directory only reads the number if the voicemail user has not
recorded his name. If the name has been recorded, it plays that,
instead.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 21 Jun 2004, Harold Workman wrote:
Just
)
ignorepat = 9
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
. Is there a
calculator or a spreadsheet which could tell me about how many calls I will
be able to make through * ?
---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
___
Asterisk-Users mailing list
able to make calls to the sip client, but after
60 secs of inactivity calls made to the client time out, due to firewall
closing nat connection.
I dont understand why i can dial to the phone, but the keepalive fails
right away. any ideas what would cause this?
---
Harold Workman
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