Displaying 20 results from an estimated 800 matches similar to: "Re Grandstream 1.0.4.38"
2004 Jan 07
3
SIP and error talking to voicemail
Hi,
I used to have a Grandstream phone connected to Asterisk a few months ago.
Worked just great!
Then today I do a new install, rather than an upgrade, and all of a sudden I
cannot check voicemail with it. No problem calling or receiving call. It
simply speeds through the vm greetings but I cannot hear them. If I check the
same VM with an analog phone it works fine.
So I wanted to check
2004 Oct 05
2
SIPphone All-in-One: coments anyone?
Hello,
can anyone comment on how one could use SIPphone's $89 All-in-One adapter
with Asterisk? Sounds to me like it should work as both a FXO and FXS.
It would be a cheap way of getting started with Asterisk and PSTN.
Any comments on the SIPphone FX200?
Any comments on SIPphone in general?
Thank you for your help
2004 Jan 09
1
Screen Pop & Remote Agents = Telemarketing
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of empire
underground
Sent: Friday, January 09, 2004 1:32 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Screen Pop & Remote Agents
> can I put a .csv file in the sql DB and have it dial from there? and
will I be able to set a
> Dial Plan to
2005 Mar 03
2
FWD and SIPPHONE problems after upgrading to CVS HEAD
I have been successfully connected (incoming and outgoing)
to FWD for a very long time. A few months ago, I changed
from SIP-based FWD service to IAX2-based, and that went fine
as well, both incoming and outgoing.
At the time, I was running Asterisk 1.0.3 Stable.
I rarely use the service, so other than noticing that I was
always successfully registered to FWD, I didn't make or
receive calls
2004 May 18
2
registering in sipphone
for inbound calls, i can register
context = from-sipphone
register => 1747xxxxxxx:passwd@proxy01.sipphone.com
but how do i configure to make outbound calls to them?
exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1)
....
[dial-sipphone]
;
; SIP to sipphone.com
;
exten => _X.,1,Dial(SIP/${EXTEN}@??????)
^^^^^^
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 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
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",
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
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
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
2004 Sep 27
1
GS 101/102 Reboot
Hi all,
Did I see on the list a while back someone creating a program to reboot the
GS phones at a pre set time.
If anyone has this program I would love a copy if it is freely
distributable.
Regards
Dave
2005 Jan 18
1
Outgoing SIP call from Asterisk problem
Hello,
I'm having a problem I can't seen to figure out. In a nut shell, I have
asterisk running with 4 accounts configured. All accounts work fine for
local calling to each other and voicemail. However, only 1 account
can make outgoing calls. All the others fail with the following error.
If anyone can shed some light on the possible problem or where to look
for more info it
2005 Feb 23
5
Zaptel Red Alarm
Guys.. I just saw this for the first time... I did some google and wiki
without any luck.. what does a red or yellow alarm mean in zaptel?
Feb 23 02:54:16 WARNING[16890]: chan_zap.c:5865 handle_init_event: Detected
alarm on channel 2: Red Alarm
Feb 23 02:54:24 NOTICE[16890]: chan_zap.c:5860 handle_init_event: Alarm
cleared on channel 2
This just happened by itself..
2004 Jan 09
3
Screen Pop & Remote Agents
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
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 Jan 19
3
configuration to Grandstream via tftp
Hi,
Anyone know how to set up tftp server for grandstream.
I gues it should be somethink like
<tftpserver-dir>
<mac-address>
firmware.bin
config.txt
Is this correct ?
And how should the config-file look like. ?
I had search sipphone.com but did'nt find anything.
/HHA
_________________________________________________________________
Rethink your
2005 Mar 23
3
Need some help
Hi all
I have a couple of questions maybe you guys can help me with them
I have sip phones , SER server , Asterisk.
what is the best way to do that (also with accounting and authentication).
which one of those options
1) sipphone -> SER -> ASTERISK -> SER -> PSTN
2) sipphone -> SER ->ASTERISK ->PSTN
on the first option i am trying to return the call to the ser