Displaying 20 results from an estimated 3000 matches similar to: "Asterisk and the IAD2431 via MGCP"
2004 Apr 12
0
*** MGCP on the menu? Check today's special!
If you're using MGCP, we need your help. There's a patch in bugs.digium.com that
needs testing by the community. Please spend some time testing and adding your
comments to the bug tracker.
The author writes:
------------------
I'm trying to make work Asterisk against a Cisco IAD2431 with chan_mgcp. Since chan_mgcp assumes the Line package is the default which is
not the usual with
2005 Mar 03
2
FWD and SIPPHONE problems after upgrading to CVS HEAD
I have been successfully connected (incoming and outgoing)
to FWD for a very long time. A few months ago, I changed
from SIP-based FWD service to IAX2-based, and that went fine
as well, both incoming and outgoing.
At the time, I was running Asterisk 1.0.3 Stable.
I rarely use the service, so other than noticing that I was
always successfully registered to FWD, I didn't make or
receive calls
2012 May 26
2
[PATCH] Update diag/mbr instruction to match the current filename.
From: Jean-Christian de Rivaz <jc at eclis.ch>
I suspect that some instructions about how to use the diag/mbr was not
updated when the source file was renamed to handoff.S. Here is a simple
proposition to fix that.
Jean-Christian de Rivaz
---
diag/mbr/README | 4 ++--
diag/mbr/handoff.S | 4 ++--
2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/diag/mbr/README
2003 Nov 18
3
"Unable to find path from G729A to ULAW" on Sipphone.com
I seem to be having a problem with transcoding and/or agreeing on a
valid codec. I am running a new image pulled from CVS at 1:30 PM CST.
The issue occurs when I try to make a call to a toll-free number over
sipphone.com.
Here's what I see in the console:
NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format):
Unable to find a path from G729A to ULAW
NOTICE[1259545280]: File
2006 Nov 13
1
Dial : Executing context/priority after bridge?
Hi,
I am using Asterisk to set up a reminder-like system, with asterisk
auto-dialing a user via SIP and playing a reminder file when the user picks
the phone. I use Gizmo service for SIP and I'm able to call through it.
However, when asterisk dials a number, Gizmo first answers then tries
bridging 2 channels. Right after answer Asterisk starts playing the
reminder.
It obviously results in
2004 May 18
2
registering in sipphone
for inbound calls, i can register
context = from-sipphone
register => 1747xxxxxxx:passwd@proxy01.sipphone.com
but how do i configure to make outbound calls to them?
exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1)
....
[dial-sipphone]
;
; SIP to sipphone.com
;
exten => _X.,1,Dial(SIP/${EXTEN}@??????)
^^^^^^
2009 Apr 26
1
1.6.1: "DNS error" but ping works
With 1.6.1 svn:
[2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout:
-- Registration for '17470121145 at proxy01.sipphone.com' timed out, trying
again (Attempt #30)
[2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable
to lookup 'proxy01.sipphone.com'
[2009-04-26 15:01:00] WARNING[1844]: chan_sip.c:10037 transmit_register:
Probably a DNS
2006 Apr 26
1
getting asterisk to reliably answer a voip line
I have a sipphone.com account, with asterisk set to
answer incoming calls, using the following settings
(phone number and password omitted) in the Peer
Details for the SIP Trunk:
allow=ulaw
context=from-pstn
dtmfmode=rfc2833
fromdomain=proxy01.sipphone.com
fromuser=1747xxxxxxx
host=proxy01.sipphone.com
insecure=very
secret=xxxxx
type=peer
username=1747xxxxxxx
The Asterisk machine is
2005 Feb 01
2
Outbound calling with TDM400P
I am trying to place an analog outbound call from a Sipura SPA-841
through a * server with a TDM400P and 4 FXO's. When I call in from an
analog line everything works fine, I can talk over the SIP phone. When
I call out, * says:
== Spawn extension (from-sip, [phonenumber], 1) exited non-zero on
'SIP/sipphone-d29d'
-- Executing Dial("SIP/sipphone-9eb0",
2005 Aug 08
4
DTMF issues with SIPPhone?
Does anyone else have DTMF issues with SIPPhone? When calling into my
DID, and entering, say, 1002. Sometimes it will recognize it properly
(rarely), other times it will receive something different. Such as,
1102 or 1000, etc. Has anyone else been having these issues? I'm
only accepting ulaw and alaw, and my relevant sip.conf information
follows:
[sipphone]
type=peer
2003 Aug 06
4
New SIP Phone
Michael Robertson, founder of both MP3.com and Lindows, has launched a
new company to supply inexpensive SIP phones ($129 for two) and related
services. See today's press release at
http://www.sipphone.com/tiki-index.php?page=SIPphone%20Inc
My question for the list is who will be the first to report on the
compatibility and usability of the SIPphone with Asterisk? The
functionality
2010 Feb 17
1
One-Way Audio after Hold
I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet,
and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet
parameters are all set correctly in sip.conf. An inbound call from Sipphone
works great until the local channel places the call on hold. During hold,
the Sipphone user cannot hear music, only silence. The silence continues
after the hold, though
2004 Oct 05
2
SIPphone All-in-One: coments anyone?
Hello,
can anyone comment on how one could use SIPphone's $89 All-in-One adapter
with Asterisk? Sounds to me like it should work as both a FXO and FXS.
It would be a cheap way of getting started with Asterisk and PSTN.
Any comments on the SIPphone FX200?
Any comments on SIPphone in general?
Thank you for your help
2003 Aug 12
1
Working with FWD, IPTel, SIPPhone?
I'll admit it. I'm a asterisk newbie (but no stranger to telephony).
The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone
specials) on a private segment calling to a Linux box acting as the
segment's firewall with a leg on our public network. The phones are
setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks
to the Asterisk HOWTO).
Getting IAX
2004 Feb 03
1
sipphone dialing out problem
Hello
when i dial a toll free no using sipphone i get this error message. How do i solve this?
Any help will be appreciated.
console message:
Starting simple switch on 'Zap/2-1'
-- Executing SetCallerID("Zap/2-1", "17473863282") in new stack
-- Executing SetCIDName("Zap/2-1", "Deepak JV") in new stack
-- Executing
2004 Apr 26
1
Problems registering with Sipphone
Has anyone else had problems registering with Sipphone over the last
few weeks?
Previously, this had worked fine. I contacted Sipphone technical
support, but they're not much help.
register => 17471234567:password@northamerica.sipphone.com/123
2003 May 29
1
a beginner's SIP question ..
I am trying to get asterisk to dial this address :
sip:723@216.52.153.207
Using a softphone on my PC (217.168.168.49)
it dials immediately and I get a voice prompt ..
I have configured an extension, 1303 on asterisk,
modifying the demo configuration :
exten => 1303,1,Dial(SIP/723@216.52.153.207)
When from my softphone I dial
sip:1303@217.168.168.51
on the console I get :
-- Executing
2005 Jan 18
1
Problem with demo on asterisk
Hi
I installed Asterisk on WhiteBox Enterprise Linux 3.0 respin 1
The process of installation was the following: First I compiled and installed
Zaptel, in order to have ztdummy (uncommented in Makefile). I loaded the
ztdummy (modprobe ztdummy) and then i installed Asterisk:
make
make install
make configuration
make samples
I started Asterisk, and created one SIP account, with the following
2017 Feb 02
2
HP Zbook17 Dock and UEFI conflict with GK107GLM aka Quadro K1100M
On Thu, Feb 2, 2017 at 4:43 PM, Phil Turmel <philip at turmel.org> wrote:
> On 02/01/2017 10:40 AM, Phil Turmel wrote:
>> Hi All,
>>
>> I've been running Gentoo on a ZBook with great success for a couple years,
>> but I've been stymied in my attempts to implement SecureBoot by an
>> apparent problem with efifb to nouveaufb handoff, but only when
2006 Oct 10
2
Xen +suse 10.1 +ASUS A6iTc w/AMD Turion Dual Core processor
Hi
I have a new ASUS A6Tc laptop with 1.6 GHz AMD Turion dual core
processor. I get a strange error:
<4>0000:00:0b.0 OHCI: BIOS handoff failed (BIOS bug ?) 000007b4
<4>0000:00:0b.1 EHCI: BIOS handoff failed (BIOS bug ?) 01010001
This only occurs when booting into the xen kernel, not without xen.
Can someone tell me why and how it can be fixed?
Art