Displaying 20 results from an estimated 4000 matches similar to: "unistim - no dial tone frequecy, no number display when dialing"
2009 Jan 24
3
Nortel IP phone i2002 - DHCP server unreachable
Is anybody using Nortel IP Phone?
I have (second hand) Nortel i2002 phone and when it boots I get:
DHCP server unreachable
F/W version: 0604D9C
My setting:
DHCP? [0-No, 1-Yes]: 1
DHCP: 0-Full, 1-Partial: 0
Can any body suggest how to troubleshoot it?
--
#Joseph
GPG KeyID: ED0E1FB7
2010 Nov 02
0
Need testing: chan_unistim improvements
Hi All,
During last three month I have worked on improving functionality of Nortel
phones working with asterisk to replace existing Nortel station by asterisk.
Many improvments done, listed below. I have only i2002 phone and unable to
test if new version of channel correctly works with i2204 phone. If anyone
can test it and report issues, it would be great. Please visit mantis to
find out patch
2005 Mar 07
3
UNISTIM channel driver available
Hello,
Cedric Hans has released an UNISTIM channel driver for asterisk (stable).
You can download it at :
http://mlkj.net/asterisk/chan_unistim-0.9.2.tar.bz2
Copy of README :
This is a channel driver for Unistim protocol. You can use at least Nortel
i2004 phones with it.
Only few features are supported : Send/Receive CallerID, Redial, SoftKeys,
SendText(), Music On Hold, Message Waiting
2005 Sep 04
0
Updated Chan Unistim?
Hi,
Does anybody have an updated Chan Unistim that compiles on Asterisk
1.2beta?
Below is the output when compiling on Red Hat 9.0
Thanks,
[root@maui2 chan_unistim-0.9.2]# make
gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g
-I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
-Wno-missing-prototypes
-Wno-missing-declarations -DCRYPTO -c -o chan_unistim.o
2009 Jan 24
0
unistim only recognize "default" context
I have in "unistim.conf
[violet]
...
context=internal
but it is not recognized. When I try to make a call it looks for context "default"
Is it a bug or a limitation of unistim.
--
#Joseph
GPG KeyID: ED0E1FB7
2008 Aug 20
2
Linksys SPA3102-NA firmware upgrade on Linux
Does anybody know if the process of upgrading firmware on "Linksys SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org
http://www.voip-info.org/wiki/view/Sipura
--
#Joseph
GPG KeyID: ED0E1FB7
2006 Feb 09
1
Unistim Packet Decoder
Anyone know of one that I could use?
Thanks,
Shri
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2003 May 13
2
Nortel i2004 IP Phones
Hello,
is it possible to connect a Nortel i2004, (i2002) IP Phone to an *
PBX?
I have seen on the net that there should be an SIP firmware aviable for
the phone, anyone an idea where to get it? Because orginally they use a
own protocol
to connect to an Meridian, BCM or CSE1K.
Another idea is to write a module for * that can handle this protocol.
bye
SLiME
2008 Nov 15
3
IAX2 client for "eee pc 1000"
What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)?
I'll eventually replace this crippled Linux with something better but I don't time to play around with it as most divers and modules are still too new and
not fully available in all distros.
--
#Joseph
GPG KeyID: ED0E1FB7
2005 May 25
2
Nortel i2004 firmware upgrade.
I've been trying to look up information on upgrading firmware on a nortel
i2004 ip phone. I have this phone leftover from a trial, and it's
supposed to be upgradable to current firmwares. Since I also run a DMS I
was able to login to nortel's site and get all the firmware files, but All
the NTP's regarding firmware upgrading these are how to tell you BCM to
send the file to it.
2008 Feb 12
3
Nortel 1140E
Anyone get the Nortel 1140E phones working with Asterisk ?
These look like great phones and I would like to start using them on
our deployments. I know these will work with Asterisk but the sample
config files are hard to find. My next step, if I cant find anything on
this list is to purchase a Nortel Communication Server for testing.
If anyone has a used NCS that works with these phone via SIP
2010 Feb 02
0
Issue when reloading
Hello list!
I?m having an issue when reloading Asterisk, I?ve had this problem in
Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same
error.
For example, I send a "reload" in Asterisk CLI and this is the output:
isb152*CLI> reload
== Parsing '/etc/asterisk/extconfig.conf': == Found
== Parsing '/etc/asterisk/manager.conf': == Found
2009 Feb 17
0
unistim channel problem
Hi
[Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type registered for 'USTM'
[Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'USTM' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
I get this after I restart my asterisk 1.6, it all worked yesterday.
I have the
2009 Feb 10
1
unistim and transfer calls
Hi
When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error?
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir:
2005 Jun 11
1
SIP-H.323 dial tone and busy tone problem.
Greetings to the list:
this is my problen when I make a call from my asterisk towards a nortel
PBX , the call is made but in my telephone sip I do not listen the dial tone
or the busy tone but the call it is completed normally.
sip-phone-g729-------------asterisk--------h323-g729--------------nortel-pbx
thi is may configuration:
RedHat 8 2.4.18-14
Asterisk 1.0.7
The NuFone
2005 Mar 24
0
chan_unistim compile failed
I am trying to compile chan_unistim and getting some compilation errors:
Asterisk Version: Asterisk CVS-03/21/05-12:24:27
AstHeaders: /usr/include
Gcc: gcc version 3.2.3 20030502 (Red Hat Linux 3.2.3-42)
OS: CentOS 3.3 (final)
The isssue seems to be stemming from AST_MUTEX_DEFINE_STATIC - here is
the error dump from chan_unistim:
gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g
2009 Jan 18
1
caller ID - handle_request_invite: Failed to authenticate user
We have a caller ID from our phone provider "Shaw Cable" (digital phone) and it was working OK until recently.
I get an error:
WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have <4>, digest has <pstn-4444>
NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user THELMA
<sip:7804789998 at 10.10.0.103>;tag=50e17675d59121c4o1
at
2008 Nov 21
2
MozIAX - Mozilla IAX2 soft-phone 3sec delay
Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone.
http://moziax.mozdev.org/
I tried it yesterday on eee pc, connected to asterisk on local LAN and the performance is terrible!
The delay is about 2sec or 3sec. and very bad echo.
I think it is the implementation of their IAX2 in their add on, as I have tried external mic. and the same delay problem.
As a comparison I've tried
2004 Oct 25
5
Nortel Phones.
Hello,
I am wondering if anyone is using the Nortel 2001 2002 or 2004 phones on
their asterisk implementation. My local dealer says they are not
compatible with any open source implementations. Is there a phone
compatibility list somewhere?
Cheers
Cian
2003 Dec 02
7
Nortel i2004
Is anyone successfully using this phone with Asterisk? There is a lot
mentioned about CISCO but nothing about Nortel...
Alex.