Please ignore my previous email. Qualify is now working well. The packets
that were misleading me were related to MWI subscription, and not the
result of Qualify (OPTIONS)..
Sorry for any confusion.
--
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I am currently not using qualify, but it seems like a nice way to know if
the phones are online. I attempted to set it up, but am running into a 404
on the subscription.
1. From the manager, Action: PJSIPNotify (with an endpoint). This caused
the following OPTIONS packet to be sent to the
working directory plus the relevant parts of
> extensions.conf. I xxx'ed out phone numbers and Google interface data.
The above tarball appears to be no longer available.
Does anyone have a copy they can put up som
possibly a result of
https://issues.asterisk.org/jira/browse/ASTERISK-26554.
Regards,
Ian
On 21/12/2016 00:01, Jerry Geis wrote:
>Hi Jerry,
> just had a look through the code, and from what I can tell, what
>you're trying to do is not supposed to work, exactly. It appears that
>what Aste
Hi,
I don’t see any SIP ACK’s in your trace.
Is the SIP 200 OK reaching the originating caller, or being blocked on
the way through?
Asterisk will tear down the call after ~30secs of audio playing in both
directions if it doesn't receive the SIP ACK.
Regards,
Ian
On 15/10/2016 12:05
0-000386f2", "DEXTEN=7077") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[s@macro-dial-one:2] Set("SIP/6010-000386f2", "DIALSTATUS_CW=") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
Hi,
PJSIP in the past had limitations on the max concurrent calls, etc. There were
ways to overcome them by changing the source code. (e.g.
http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-February/015721.html
The following bash 1-liner may be useful...
while true; do top -Hbc -p `pgrep asterisk` -n 1 && asterisk -rx "core show
threads"; sleep 1; done
Regards,
Ian
On 24/07/2016 13:39, Tzafrir Cohen wrote:
>
> On Fri, Jul 22, 2016 at 12:02:43AM +0100, Chirag Desai wrote:
&
also had no effect.
I would appreciate any suggestions for troubleshooting or things to try.
Thank you!
- Ian
--
Ian Harding
IT Director
Brown Paper Tickets
1-800-838-3006 ext 7186
http://www.brownpapertickets.com
--
_
-- B
016, Ian Harding wrote:
>
>> Inbound route: Don't Care
>> Queue: Yes
>> Extension: Don't Care
>
> What front end are you using?
>
> What version of Asterisk, OS, etc?
>
> You may get more interest on a mailing list specific to that front end.
>
--
Ian
and tried searching with
Google, but I haven't been able to find a clear answer.
This is Asterisk 1.8.20.0, BTW.
Thanks!
--
Ian Pilcher arequip...@gmail.com
Sometimes there's nothing
set debug ip the ip address of your VoIP provider
Can this be set in a config file?
This is a sporadic problem, so I need to set the server up to log the
appropriate information when it occurs.
Thanks!
--
Ian Pilcher
.
--
Ian Pilcher arequip...@gmail.com
Sometimes there's nothing left to do but crash and burn...or die trying
was unable to cleanly stop BIND on
this box. (32-bit Scientific Linux 6.1, BTW.)
Anyone have any idea what just happened?
TIA!
--
Ian Pilcher arequip...@gmail.com
If you're going to shift my
).
--
Ian Pilcher arequip...@gmail.com
If you're going to shift my paradigm ... at least buy me dinner first
before I got to the part about
changing the Apache user ID!)
--
Ian Pilcher arequip...@gmail.com
If you're going to shift my paradigm ... at least buy me dinner first
!
--
Ian Pilcher arequip...@gmail.com
If you're going to shift my paradigm ... at least buy me dinner first
--
Ian Pilcher arequip...@gmail.com
If you're going to shift my paradigm ... at least buy me dinner first
I've no experience with that phone model or protocol. But if you run a tftp
trace you'll see what files the phone is looking for.
Check my old thread on pbxinaflash forums for details.
i
-- Original Message --
Received: 04:59 AM COT, 06/16/2011
From: bilal ghayyad bilmar...@yahoo.com
no from tag, dropping callid:
00078599-2e4e0002-23aa7a4e-0b32ceef@192.168.1.114 from:
sip:702@192.168.1.41;user=phone
[2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:24110 handle_request_do: Invalid
SIP message - rejected , no callid, len 337
ian
...
-- Original Message --
Received: 07:31 AM
-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3
On Mon, May 30, 2011 at 2:45 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
Console is showing the following. Looks like it doesn't like the format of
the
REGISTER message
--
Received: 05:11 PM COT, 05/30/2011
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3
On Mon, May 30, 2011 at 5:18 PM, Ian S. Worthington
\000\000\000\000\002\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000
...
Tried changing, as per your suggestion, to nat=yes and your given settings in
both SIPDefault.cnf *and* SIPnncnf without change.
ian
And f/w POS3-07-4-00
i
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I am having a problem registering my cisco phones which is exactly like that
described in
http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html
except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00
The symptoms are:
o 7960 lines show [X]
o Outbound calls can be
-- Original Message --
Received: 03:45 PM COT, 05/28/2011
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3
On Sat, May 28, 2011 at 4:08 PM, Ian
quickly establish whether or not this is the case.)
--
Ian Pilcher arequip...@gmail.com
this situation would be appreciated.
Thanks!
--
Ian Pilcher arequip...@gmail.com
this working
on a recent Fedora release (using the Festival application, not AGI).
Thanks!
--
Ian Pilcher arequip...@gmail.com
I had a system running on Xen in test. I had terrible echo problems with a
SPA3000. As a reference, I swapped to bare metal machine and although I still
had echoing, the echoing was much closer to the original sound. The Xen server
was idle apart from the AsteriskNOW installation. So, this lead
Forgive the possibly stupid question, but do these problems you describe
apply equally to the dom0 as to any domU's in a xen system? I used to
think not, but now I'm starting to realize that I'm probably mistaken...
Dom0 is still a virtual machine, so I would say so.
--
On 09/16/2009 08:53 AM, Jared Smith wrote:
Please open a report on our issue tracker at http://issues.asterisk.org/
Will do. Thanks!
--
Ian Pilcher arequip...@gmail.com
notifications to both
extensions, etc.
Anyone seen anything like this? Known limitation/bug?
Thanks!
--
Ian Pilcher arequip...@gmail.com
confBridge in Asterisk 1.6.2.0-rc1 doesn't stable.
It causes segment fault very often and results in asterisk crash.
Ian
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Register
Same here, it doesn't remain unreachable for long, but it's annoying if
a fax arrives when it is. I'm going to try Fax for Asterisk instead:
http://www.digium.com/en/products/software/faxforasterisk.php
Ian
Marco a écrit :
Hi All,
I'm having a strange problem and I'm not able to understand
Sorry for the stupid question, but I think I'm not understanding
something. Why can't you use Fax for Asterisk with res_fax and
res_fax_digium?
Florian Hackenberger a écrit :
Thanks for the explanation! Sounds all good. There is one remaining
question however. As you mentioned T.30, is
Thanks to everyone who replied. I'll let you know if I have any issues
with Fax for Asterisk...
Regards
Ian
Hello
I am going to try the new Digium Fax for Asterisk product. I'm planning
to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs.
I'm looking at Grandstream
be
grateful if you could share your experiences.
Regards
Ian
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Thanks for that, I'll add Audiocodes to the list. By the way, we usually
buy our VoIP kit from VoIP Supply - we've always been very satisfied :-)
Regards
Ian
Cory Andrews a écrit :
Both of those handle fax. In my experience Audiocodes works best for fax.
Their MP202, MP203 models are rock
Beware PoE switches that can't handle Class 3 (15W) on all ports.
Most have fans because 24 (or 48) x 15W is hot!
IanC
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: Sunday, February 01,
to deny a user to call
if you need any config files I will send them, seeing as I dont know what
files to send.
If you can point me to a howto I will be more gratefull.
Regards
Ian
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Also try putting Asterisk in the audiopath by setting canreinvite=no in
sip.conf
Regards
Ian
On Sat, Jun 7, 2008 at 4:07 PM, Michiel van Baak [EMAIL PROTECTED]
wrote:
On 08:36, Sat 07 Jun 08, Russell Bryant wrote:
On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote:
i have this on my
direct me to the right site I would be realy greatfull.
If you need any more info, I will be only to happy to provide it, the
piece of my dialplan is below.
Thanks in advance
Regards
Ian
exten = 9876,1,Progress()
exten = 9876,n,dial(ZAP/1/0720311294,,m(default))
exten = 9876,n,Hangup
bumped the txgain up a notch a while back, can it be because of this?
I ran a top and saw that the server only have about 16Mb free ram, can
this be a possible cause?
My zapata.conf and zaptel.conf are below.
Thanks in advance
Ian
# less /etc/zaptel.conf
# Autogenerated by /usr/sbin/zapconf on Fri
info, I will be only to happy to provide it, the
piece of my dialplan is below.
Thanks in advance
Regards
Ian
exten = 9876,1,Progress()
exten = 9876,n,dial(ZAP/1/0720311294,,m(default))
exten = 9876,n,Hangup(
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Hi Raul
Raúl Gómez C. said the following on 05-Mar-08 07:40 PM:
Ian,
I'm unable to transfer calls using *2, I'm not sure why. Here's my
configs:
snip
In the phones the /Send DTMF:/ is set to in-audio and via SIP INFO
It should only be set to SIP INFO, or else the audio comes out too
.
Ian
Raúl Gómez C. said the following on 04-Mar-08 03:33 PM:
Hi Ian,
I will try this workaround, I'll be trying to get this to work with
the chan_local solution, if I have success I'll let you know, thanks...
On Wed, Mar 5, 2008 at 2:37 AM, Ian [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED
Hi Raul
I have bypassed my Grandstream's transfer function, by enabling *2
transfers in features.conf, and setting canreinvite=no in sip.conf
Hope this helps you
Ian
Raúl Gómez C. said the following on 03-Mar-08 08:34 PM:
In the config file (sample) features.conf are some commented
:
Ian,
I'm having *THE SAME PROBLEM* and I've noticed that when a transfer
fail (only happens when receptionist dial an external number) the call
is marker as NO ANSWER in the CDR, even when the call *HAS BEEN
ANSWERED* by the other party (the callee). See my previous post below.
http
Gordon Henderson said the following on 25-Feb-08 10:26 AM:
On Mon, 25 Feb 2008, Ian wrote:
Mojo with Horan Company, LLC said the following on 22-Feb-08 07:58 PM:
Sorry, I jut got your other message stating the steps your boss' secretary
uses to transfer calls, so this question's time
manuals of the phones, and the 'flash' key is the only way to do an
attended transfer, I will try the keys defined in features.conf today to
see if it makes a huge difference, is there any funny configurations I
should be aware of before I start playing around with the features.conf?
Thanks
Ian
phone.
The big thing about this problem is that it comes and goes, like
yesterday we didn't have a problem, and I did not change a thing.
Ian
Ian wrote:
Hi All
Sorry to be a bother again but seems like I just cant get away from
the problems.
This time my problem is that *sometimes
=info
callgroup=1
pickupgroup=1
call-limit=20
subscribecontext=GXP_BLF
canreinvite=yes
nat=no
[300](sets) ;Luzaan
regexten=300
Any help will be apreciated
Thanks
Ian
Ian said the following on 22-Feb-08 10:06 AM:
Hi,
Mojo with Horan Company, LLC said the following on 20-Feb-08 09:31 PM
.
Thanks in advance
Ian
--
www.vddi.co.za http://www.vddi.co.za/
I Coetzee
IT Tegnikus
Telefoon: 012 664 2300
Selfoon : 079 522 6519
Faks: 012 644 2902
E-pos : [EMAIL PROTECTED]
Skype : vddb_igcoetzee
to have to only recompile zaptel, or is that
the way of doing things?
Thank you all for your support
Please scroll down to see the answers to my own stupid questions :-)
Regards
Ian
Ian said the following on 04-Feb-08 09:38 AM:
Thanks for the speedy reply
Tzafrir Cohen said the following on 30
Hi
Thanks for the response
Anthony Messina said the following on 01-Feb-08 03:36 PM:
On Thursday 31 January 2008 11:52:09 pm Ian wrote:
Sorry for taking so long to reply,
This email got lost in translation, again.
Ian
Ian said the following on 30-Jan-08 03:57 PM
Thaks
Thanks for the speedy reply
Tzafrir Cohen said the following on 30-Jan-08 12:37 PM:
On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote:
Hi all
I have a small problem here. I asked this question on another asterisk
mailing list, but nobody seemed to be able to help me there.
We
Sorry for taking so long to reply,
This email got lost in translation, again.
Ian
Ian said the following on 30-Jan-08 03:57 PM
Thaks for the speedy reply
Tzafrir Cohen said the following on 30-Jan-08 12:37 PM:
On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote:
Hi all
I have a small
Just testing to see if my emails to this mailing list gets through.
Tried posting a question, but it failed
Thanks
Ian
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format,
which is available for download at
http://www.iancoetzee.za.net/tone_dial.mp3. After listening to the
playback I concluded that the DTMF signals being sent is totally wrong.
The relevant pieces of my configs are below
Your help in this matter will be greatly apreciated.
Regards
Ian
be
implimented in the PBX, not the phone.
Maybe have a look at Mitel, you can tickle a URL on the phone to
make it dial, so clicking on the name on the company directory in
the intranet will call them using your phone.
Ian
--
Ian Freislich
simple, so please do not worry about stating
what may seem to be obvious. Thanks in advance for your help.
Ian.
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: 03 June 2007 18:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Options Reply Ignored
On Sun, 3 Jun 2007, Ian Clough wrote
and a
retransmission is made and the phone is marked as UNREACHABLE and will not
accept any calls.
Any ideas?
Ian C
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). Is it zaptel driver
issue? The server loading actually is not so high before crash.
I've been investigating this issue for weeks and I'm totally out of ideas,
so any help or suggestions anyone could provide would be greatly
appreciated…
Best regards
Ian
The following is my server configuration detail
:
to='gmail.com'./str:text/stream:error/stream:stream
So does this only work if you have email accounts from gmail.com?
Thanks
Ian.
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in
the sip.conf but need this to be dynamic based.
Thanks
Ian.
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Flags: bus master, medium devsel, latency 64, IRQ 217
I/O ports at de00 [size=256]
Memory at fe9fe000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
Can anyone help?
Thanks
Ian
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Anyone had any luck getting the Unistim channel driver to install on
Tribox 1.0.5?
Ian Cowley
Network Security
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This email has been scanned by the MessageLabs Email Security System.
For more information please visit http
Anybody have suggestions on having a 56K dialpool and VOIP
connections with an Asterisk box over the same set of PRIs? We've
done the PM3 with PRIs for just dialup, but are looking for a way to
integrate our Asterisk box and move our voice calls onto the same PRIs.
Ian
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.
Regards,
Ian.
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Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension 1055.
Outbound calls to 1055 work perfectly.
Inbound calls
-Original Message-
From: Ian Cowley
Sent: 27 January 2006 15:10
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] External IAX2 phone defined as internal
behaving as from PSTN
Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg
)
Sent: 27 January 2006 15:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] External IAX2 phone defined as internal
behaving as from PSTN
On Fri, January 27, 2006 16:09, Ian Cowley said:
Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg
.
Is this configurable?
Ian Cowley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian Cowley
Sent: 27 January 2006 15:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] External IAX2 phone defined as internal
behavingas
, Ian Cowley said:
Iax.conf
[general]
;bindport = 4569 ; Port to bind to (IAX is 4569)
bindport = 5036 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=g729 ; 4 simultaneous allowed
allow ilbc
Make sure you have a recent copy of the firmware. There was a bug
preventing registrations from succeeding until Nov 08 2005 and newer
firmwares.
--
Ian White
South Island Community Access Network (SICAN)
email: [EMAIL PROTECTED]
http://sican.tc.ca/
On Dec 13, 2005, at 17:24, [EMAIL
I've been testing various phones with [EMAIL PROTECTED] 1.5
So far I like the Grandstream GXP-2000 for price and ease of
configuration, though SideTone/ComfortNoise only comes with the current
Beta code
I like the SNOM 320 buts its more expensive.
Its more of a business phone and doesn't have
, using *, I
always get an error 'Cannot decode callerid'.
Does anyone know if I need to patch the vpb driver or something?
Many many thanks,
Ian Bonham
P.S. the POTS I'm connected to is British Telecom's SystemX if thats any
help
Thanks John.
I can't seem to see if just applying the Asterisk side of the fix will
correct things though. The card I'm using is a VoiceTronix OpenSwitch 12.
I'm using the vpb driver as opposed to the Digium drivers in this instance.
Any clues?
Thanks,
Ian
From: John Crowhurst [EMAIL
and the vpbhp driver.
The echo occurs on both SIP-POTS calls, and SIP-SIP calls. I've tried a
number of volume adjustments to correct the echo but it is always the same.
If anyone has any ideas I'd really appriciate some help, as this is a major
urgency,
Many many thanks,
Ian Bonham
I'm using a Voicetronix OpenSwitch12 analouge card. It's on the latest
driver for Asterisk (2.4.9) and has echo cancellation turned on. This works
fairly well on on SIP-POTS calls after it trains up, but there is still a
small echo. The SIP-SIP calls are really echoy though.
Cheers,
Ian
Hi Arne,
In /etc/asterisk/voicemail.conf, under the [default] section, you need to
declare the users like this :
box# = passnumber for box, Name of User,email address
e.g.
221 = 1234,Ian Bonham,[EMAIL PROTECTED]
Do that for each mailbox you require.
Then in the sources directory, under
Not sure about the Digium, but I can tell you +34 is Spain, if that helps
you track anything down? I assume you've tested the line with a normal phone
to make sure it's not a telco fault?
Ian
From: Angus Comber [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
, but wonder if there is a
gain setting in * it's self I may have missed?
Cheers for your help,
Ian
From: Matt [EMAIL PROTECTED]
Reply-To: Matt [EMAIL PROTECTED],Asterisk Users Mailing List -
Non-Commercial Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non
line or just the SystemX dial tone, as though the vpb has opened
the line, but not sent the DTMF.
If anyone can help I would really appriciate it,
Kind Regards,
Ian Bonham
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Hi all, just joined the list.
Wanted to say 'Hi'. I'm an Asterisk user in Gibraltar, and wanted to say
'Hullo' to you all. I'll watch the list and I if I can help out anywhere I
will.
I'm running Asterisk on a RedHat9.0 system, with a VoiceTronix OpenSwitch12
card (sorry Mark!) and I really
Can anyone help me how to open recorded converstations in asterisk?
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I'm new to asterisk. I would like to ask if there's a feature in
asterisk wherein you can monitor ongoing calls, some kinda like
tapping into active phone calls? It must have this feature but I do
not know where to get some reference to set this up or test this.
Can anyone share me some sites as
work with
this? I've not
seen any zaptel.conf that supports this. Any workarounds?
Thanks for any help!
Ian
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I have a modem and my digium card in my PC. The problem is when i try
to establish an incoming call, the modem responds first before the
digium card. Is there any way to allow asterisk to get the call first
before my modem?
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On Jul 6, 2005, at 9:07, Kevin P. Fleming wrote:
Ian White wrote:
The use of the nonce looks right to me. Can somebody point out
what is going wrong here?
Yes, I agree, it looks correct. However, what version of Asterisk
are you testing against? Current CVS HEAD adds 'stale=true
to connect to Asterisk Manager (111)
Warning: fclose(): supplied argument is not a valid stream resource in
/var/www/admin/functions.php on line 2349
thnx,
ian
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I've just Installed [EMAIL PROTECTED] i browsed it's built-in AMP. it
prompts for a login if you click on asterisk management portal. i
tried
user:[EMAIL PROTECTED]
pass:password
and
user:admin
pass:password
but it didnt get through. do you the default login for it?
thnx,
ian
The use of the nonce looks right to me. Can somebody point out what
is going wrong here?
Jul 4 14:37:04 VERBOSE[12919]:
Sip read:
REGISTER sip:voip.victoria.tc.ca SIP/2.0
Via: SIP/2.0/UDP 199.60.222.229:5060;branch=z9hG4bKUF2CCx
Max-Forwards: 70
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how can i solve the error on the last part?
need help. thnx...
Zaptel Configuration
==
SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03:
I guess I need to go over my college electronics again... it's been a while.
Ian
[EMAIL PROTECTED] 24/05/2005 14:18
Yes, one might think that, IF one didn't understand the nature of
electricity and electrical components.
SOME people also puzzle over the fact that you can't boil eggs
it without being destroyed?
I've tried sever hard and soft resets to revive the module, reseated the card
and the module and moved slots to no avail. Can anyone thing of anything else
to try?
Thanks,
Ian
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416
trigger
some phones.
Incidentally I have a SPA-1001 and it's giving me no problems.
Ian
[EMAIL PROTECTED] 10/05/2005 17:52
I hooked up a SPA-1001 with asterisk yesterday and all works well except
the phone doesn't ring.
The phone I'm using has a LCD display so I can see the call come
I have a cellsocket working with a Nokia 6150 right now. Funny the
model I bought was a cellsocket for a Nokia 5110, and for some reason,
it wont work with the 5110 unit i put in. The 6150 works like a
charm, though. For some reason, it ignores the first two characters
of the phone number you
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