Displaying 20 results from an estimated 10000 matches similar to: "Issues"
2006 Dec 05
4
Attended Transfer
Dear List,
I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable
attended transfer feature. but i just can't do it work. I've already
set "atxfer = *" (and many other combinations) and all extensions on
extensions.conf have the t and T option. But when I'm going to test,
it doesn't work. Is there any other file that i have to configure in
order to
2008 Apr 30
0
AVAYA 8300 integration with asterisk 1.2.x
Hi All,
I need help with integrating AVAYA 8300, the avaya can do outbound
calls but cannot do inbound calls, im sending calls from sip to avaya
using E1 ISDN line. My config was based on aspect dialer it's working
with aspect but not with avaya.
My config and error is below.
zaptel.conf
span=1,1,1,ccs,hdb3
bchan=1-15,17-31
dchan=16
zapata.conf
group=0
context=avaya
switchtype=euroisdn
2005 Jan 11
2
SIP, * and clients behind NAT
I am new to VOIP, Linux and Asterisk. Through a lot of reading (this
list, voip-info.org, documentation, etc.), I successfully installed FC3
and * on a new Dell SC420 with two X100P connecting to two PSTN lines at
my office. I've also installed AMP to help me configure IVRs, call
groups, extensions, etc.
I use a Handytone-286 ATA and x-lite clients on the internal network and
all works
2004 Mar 06
2
GS HandyTone-286 Transfer Problem, can anyone confirm?
There seems to be a problem related to the Grandstream HandyTone-286.
When a call is placed through the adapter, the call can be
transferred. However, when a call is received through the adapter,
the call cannot be transferred. The problem does not exist with a
BudgeTone-101 (1.0.4.23) using the same Asterisk configuration and
Dial() settings (Ttm). I tried all of the firmware on their BETA
2005 Mar 24
0
Native Bridging drops call on release
Has anyone experienced a dropped call when bridging? I get an "OK,
ready to transfer" from both channels, but when asterisk releases the
call, it is dropped immediately by the upstream provider. I've tested
against another provider and it works fine, and it also works fine
across two different providers, including TO and FROM the one that's
acting buggy.
Here's a
2004 Sep 01
0
TDM40B hangup on fax or data modem carrier
Hi !
I have a TDM40B and i try to use it connected to modem for incoming call
data transfert.
I have no problem to use it with a phone and a talk communication work fine.
But when we try to use with modem, with most modem, we got data carrier for
few seconds and channel hungup.
< [ TYPE: Null Frame (4) SUBCLASS: N/A (3) ] [Zap/4-1]
-- Zap/4-1 is ringing
<< [ TYPE: Null Frame
2003 Nov 25
3
Handytone 286 - calling out
Hi,
Just received recently released Grandstream handytone 286 ATA for
testing.
I can call ATA from any other extensions and conversations seems to be
of quite good quality. However placing calls from ATA is not possible at
all to any extensions.
After dialing there no indications of any kind from ATA at all. It just
"hangs in there".
ATA is behind NAT, registers to an * with public IP
2004 Jan 04
8
Grandstream Handytone 286 RTP Problems
I am trying to get the handytone 286 to make a very simple call to * and
having problems. It registers with * just fine, but when I place a call
(to echo test, for example), the RTP stream seems to have problems
opening. Here is there error I get in *:
WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for
2017 May 08
2
Second DC won't start LDAP daemon
Hello.
I've got a network of FreeBSD servers which traditionally hosted a
classic domain.
I upgraded some months ago, removing the old PDC and BDC and migrating
to an AD DC controller in a jail.
This is working fine with Samba 4.4.13.
Now I'm trying to add a second DC, so I created a new jail on another
physical server and went on with the setup, following:
>
2003 Nov 17
1
SIP calls no longer work
Hello,
I'm having a problem with SIP. More specifically, I
can't make any calls using SIP.
I have had an iConnectHere account and Free World
Dialup account working for quite some time, and now
all of a sudden I can't make any SIP outgoing calls.
PBX*CLI> sip show registry
Host Username Refresh State
192.246.69.223:5060 XXXXX 120 Registered
2005 Mar 10
5
asterisk and Broadvoice Outgoing Again :(
Hi,
I can't make outgoing calls via Broadvoice. I have tried each and every
configuration that was posted to list previously.
I am able to receive incoming calls fine.
I get the following in asterisk console:
=====================================================
asterisk*CLI> show version
Asterisk CVS-HEAD-03/10/05-22:51:28 built by vicky@asterisk on a i686 running
Linux
2004 Aug 27
1
Help with a fax via Grandstream Handytone 286?
I have an analog Fax machine which I wish to connect to the network and
the Asterisk server. It will connect through a GS Handytone 286
converter and then into the LAN. Is there any information out there on
what I need to write in *sip.conf* and/or *extensions.conf* to make sure
the fax works as a fax?
Channel 8 on my T1 is a reserved, dedicated line for the fax number. Do
I need to
2004 Dec 01
0
Grandstream BT100 / HandyTone 286 and Level 3
Hello,
Has anyone gotten a Grandstream BT100 to work with Level 3's 3Tone?
I've been able to get my extension to interface with it, but there is no
sound
and the call on the GS side terminates prematurely.
Here is the relavent portion of the SIP.CONF
[4007] ; Budgetone BT100
type=friend
insecure=yes
context=test-budget
username=4007
fromuser=4007
callerid=4007
host=dynamic
nat=yes
2005 Aug 11
0
* behind NAT, client behind NAT(handytone 286), very strange behavior
Hi All,
I've an Asterisk Server behind a NAT.
Using DNAT, I've opened port 5060 and all 10000:20000 udp.
Sip configured with externalip and subnet.
I've another site, also with NAT, where I map the rtp port (as defined
in the client) to map to the local client (DNAT).
Using Xlite, this configuration works, it requires using the quality=yes
and NAT=yes/always in the sip ext
2003 Oct 26
1
NuFone International Calls
Does anybody know how to do an international call using NuFone. I realise
this isn't really the place to ask, but NuFone appears to be closed for
the weekend and would like to have a try at this before tomorrow. I
assumed it would be '011' for an international line followed by country
code but that doesn't seem to work.
I am getting:
-- Executing
2004 Nov 24
2
Graststream ATA 286 Caller ID Europe
Somone in europe have had succes getting Callir ID showed on a phone
screen conected to an Handytone 286 ?
Adri? Vidal
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2005 Feb 02
0
ZAPHFC Drop calls
Hi everybody,
I had an ISDN card with winbond chipset and isdn4linux, but there was a lot of
echo when call from SIP to ISDN, so I buy an HFC chipset card (Conceptronic
c128i). I downloaded and compiled bristuff-0.2.0-RC5 and everything is going
fine. The sound quality is excellent, there is no echo, but i have a strange
problem. Sometimes calls made from SIP phones to ISDN just finished
2005 Feb 02
9
911 and Cops knocking on my door
Hi,
I am quite new to asterisk so I am not sure what is needed to figure
out this problem. If more information is needed and not provided I
will gladly provide it.
I have a very basic asterisk setup. 1 x100p card and a grandstream
handytone 286. I can make calls fine to most phone numbers from the
handytone device the trouble seems to come when I dial this number
591-1079. It puts me through to
2004 Jul 07
0
Audio cuts off 10 minutes into calls
Hello list,
We run Asterisk CVS-HEAD-06/02/04-11:25:18 built by root@Gate01 on a i686
running Linux.
All works fine except Audio is lost 10minutes into the call. This happens
for every call
PSTN-SIP, SIP-PSTN, SIP-SIP
Example of one call setup using Snom200 and Grandstream 486:
-- Executing Macro("SIP/xxxx1251-d638", "CFW|xxxx1251|SIP/xxxx1253") in new
stack
-- Executing
2010 Apr 29
4
ATA shootout: PAP2T versus Grandstream Handytone 286
I'm considering a situation where I buy about twenty ATA devices.
I've played with the Linksys / Cisco PAP2T, and got that working fine
with some inbound and outbound faxing. The web GUI was okay. I'm
seeing prices around $45 to $50 for this thing. It comes with two FXS
ports, but I only need one FXS.
I've seen the Grandstream Handytone 286 online. It looks promising as
an