Displaying 20 results from an estimated 2000 matches similar to: "Running Asterisk and NAT on the same box?"
2016 May 23
6
Wildcard X100P Disconnect Problems
Hi All,
When the Caller hangup at the voice menu, the wildcard X100P didn't
disconnect the calls properly and it just keep looping at the voice menu and
timeout and loop again, are there any methods can fix the problems?
Please help!
Thanks,
Randal
2003 Nov 21
9
Outline For Asterisk Book - Please Review & Comment
Asterisk Users
In an attempt to help Asterisk move forward, a number of us have decided
to create a book. It would initially be released as an "ebook" that
could be sent to newbies to help them up the rather steep learning
curve. Ultimately I would like to see it published and sold in
bookstores (preferably by O'Reilly & Co.).
Below is the outline for the book. We REALLY
2003 May 19
1
Call between G.711 and GSM
*This message was transferred with a trial version of CommuniGate(tm) Pro*
Will asterisk actually convert between two different codecs?????
ie, a SIP endpoint running GSM and another running G.711?
Wouldn't that add quite some latency? I was always under the impression
Asterisk did not recompress and was smart enough to negotiate the right
codec at each end and just pass through the RTP
2003 Jun 17
11
Speex
Hello everyone.
I am having problems getting speex support.
It seems * is not loading speex. When i did a make in the codecs sub dir,
the following error pops up when making speex:
codec_speex.c:34:19: speex.h: No such file or directory
is this file missing in the cvs as i just removed the whole * dir and did a
new checkout and still seem to get this error, or do i need to get/install
2003 Sep 22
2
G.729A + Cisco AS5300
Hello,
I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected.
The codec list show on my cisco AS5300 for g.729 are:
g729r8
g729br8
I suspect that
2003 Nov 21
5
Asterisk Call Manager for Windows 0.0.1 (Alpha)
If anybody is interested, I have an early version of my Call Manager for
Windows application integrated with Asterisk. CMW is an application bar
(like the task-bar) that docks to the top of your desktop window. It
provides the following functions:
1. View Call-Related Information (Caller ID, Call State, Call
Direction)
2. Monitor Status of Asterisk Stations (Channels) -- BLF or "Busy
2003 Dec 02
5
Iax Client Library Issues? (DIAX, iaxComm, etc.)
Hi,
I seem to be having problems with IAX clients based on the iaxClient
library. I have been working on my own client (an augmentation to the
Call Manager I released last week) and it seems to regularly miss
incoming calls entirely. It also occasionally misses the drop signal
when the remote end drops a call.
Has anybody else seen this kind of behavior? I have tested with my
client, with
2005 Jun 27
8
OT: Good soft-phone on Linux
Hi Folks,
I am wanting advise on a good soft-phone on Linux. I have looked at
Gnophone but cannot seem to get it to compile under debian sarge. I am
now looing at sipXphone seem to be picking up that it is not that
stable, but perhaps someone here can advise on what softphone I can use
on Linux.
Thanks in advance,
Hamish
-------------------------------------------------------------------
|
2003 Sep 12
5
Asterisk using a h323 gateway
Hello:
I am testing Asterisk with oh323.
My question is: can Asterisk route some calls thru a second h323 gateway (a
h323 <-> PSTN gw)?
- Asterisk ip: 192.168.1.10
- h323<->PSTN gw: 192.168.1.20
I've tried:
exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20)
or
exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20)
but it does not work at all.
If my h323 client
2004 Jan 21
9
New Windows IAX Client
Announcing a new Windows-based IAX/IAX2 client. Please download it and
give it a try. Let me know about any bugs, and any missing features. I
have yet to come up with a catchy name for it, so at this point it calls
itself IAX Phone. (Suggestions? Non-derogatory suggestions,
preferably).
Download: http://www.sokol-associates.com/Downloads/IaxPhone.zip
Reference & Support Page:
2004 Apr 02
3
WiSIP Firmware Version F?
Greetings,
I purchased a WiSIP at the VON conference and am now trying to configure it
to work with Asterisk. I have read all of the previous postings regarding
the WiSIP and most of the information apparently does not apply to the
version of firmware installed on my phone (version WF.00.0F).
I cannot get the WiSIP to register with my Asterisk box. It leases an IP
from my DHCP server, then
2003 Oct 14
4
Asterisk capacity
Hi,
I am really interested in the true capacity that Asterisk can handle.
What is the maximum number of users that can be handled by Asterisk on a
standard 2.4G P4 IBM server or similar? Anyone has a clue?
Cheer,
Vincent
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2003 Jun 15
2
Voicemail with H.323?
Trying to configure voicemail with H.323 all I get is the following errors
when I call 123, 666, 665, 664 or 031. I'm a newbie at this so, I think it
might be a simple fix.
[chan_oh323.so] => (OpenH323 Channel Driver)
== Parsing '/etc/asterisk/oh323.conf': Found
0:00.004 OpenH323 Wrapper OpenH323 Wrapper Version
0.0alpha0 by inAccess Networks
2003 Jun 12
4
Voicemail message as e-mail attachment
Hi all,
There is something special I must configure in order to get the voice mssage
by mail?
In voicemail.conf I have:
serveremail=asterisk@mydomain.ro
attach=yes
[default]
301 => 6535,Home Mailbox,dtoma@fx.ro
I have tried to let a message for 301, but this message is not forwarded by
mail.
I am missing something?
Thanks,
Dan
2011 Apr 13
1
Asterisk Tech Tips: Cookin' with Asterisk
Greetings Asterisk Users,
I'm happy to announce that Russell Bryant and Leif Madsen have volunteered to host the next Asterisk Tech Tips webinar, next Thursday April 21 at noon central time. Russell and Leif are project leaders and have collaborated on two Asterisk books: Asterisk: The Definitive Guide and Asterisk Cookbook , both published by O'Reilly & Associates. Asterisk: The
2003 Oct 05
2
Good W2K softphone
Hi
U can visit the http://iaxclient.sf.net for some opensource underdevelopment
softphones.
Take Care
Obaid Amin Syed
>From: Chris Albertson <chrisalbertson90278@yahoo.com>
>Reply-To: asterisk-users@lists.digium.com
>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] Good W2K softphone
>Date: Fri, 3 Oct 2003 23:00:13 -0700 (PDT)
>
>
>I haven't
2003 Nov 03
2
Transfer from Grandstream BT100?
Hi,
Does anybody know how to properly execute a transfer (without using the
|Tt option) from a GS100? Scenario:
1. I call from X-PRO (ext 1100) to Grandstream (1101).
2. Grandstream answers. Call is established.
3. Press [TRANSFER] on the Grandstream. X-PRO caller is put on hold.
Grandstream gets dial tone.
4. Grandstream dials 1103 (the extension of another GS100).
5. Grandstream hangs
2004 Jun 27
1
IAX Phone Issues/McAfee Virus Scan vs. IAX Phone
--Request For Bug Reports--
I'm working on the next release of IAX Phone. Please let me know what, if
any, issues you who use it may have run into. I hope to be able to release
a new version in the next two weeks.
Some fixes/features:
- Conferencing
- Proper handling of 'qualify'
- Intercom
- Paging
- Phone Book
--Virus Scanner Problems?--
I have been working through a number
2003 Apr 17
4
Xten / SIP Phones compared to GnoPhones
I have seen a couple of messages on the Xten and the work done by William
Walsh (Kudos). It is not clear in my mind the advantages of SIP phones
versus using GnoPhones (once we complete the work for the Windows version).
Since I lack the experience with IP SIP phones, can someone, high level,
tell me when it makes sense to use them. Is it complicated to set up on the
Asterisk side?
Thank you.
2003 Oct 22
0
SV: Running Asterisk and NAT on the same box?
Hi
I'm running exactly the same setup. Asterisk is running on my FW/NAT/Router
with two interfaces. My local phones are situated behind the NAT and connects
to the outer interface of the */FW/NAT/Router. * is then connected to my
SIP providers (since I'm only using the SIP-part of *, PSTN connection through
my SIP-provider). Works fine!
rgds,
/staffan kerker
sweden
-----Ursprungligt