Hi,
Has anybody come across a situation where they were unable to register with
Asterisk through a SIP stateless proxy server?
I'm getting an error:
403 Authentication user name does not match account name
As far as I can tell the requests reaching Asterisk with and without the proxy
are
-Original Message-
From: Pedro Nunes [mailto:[EMAIL PROTECTED]
Sent: 15 December 2005 08:59
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Starting RTP with Dial and MusicOnHold
Hello,
Do you try
Answer() and
-Original Message-
From: Elton Machado [mailto:[EMAIL PROTECTED]
Sent: 15 December 2005 14:03
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Starting RTP with Dial and MusicOnHold
Why not to use r option in
Hi,
Started getting a bombardment of these messages on the Asterisk console this
morning (20+ a second):
Dec 14 10:00:30 WARNING[17006]: channel.c:588 ast_queue_frame:
Exceptionally long queue length queuing to SIP/bluecity29-a5cfDec 14
10:00:30 WARNING[17006]: channel.c:603 ast_queue_frame:
Hi,
I'm trying to get Asterisk working with a supplier's Cerpack switch and
everything is working except audio ringback for calls coming from Cerpack to
Asterisk.
The Cerpack switch only does out of band progress indication (seems a bit
strange for SIP to SIP calls?!) so I've spent the last two
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Aaron Clauson
Sent: 24 November 2005 03:07
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Satellite, SIP Invite 488 Codec
Rejection,SIP Timing Issue??
Hi,
Thanks for the tip
Hi,
I have a very strange Asterisk SIP call signalling problem that is proving
extremely difficult to track down. The problem is that any SIP INVITE
request that is coming into Asterisk over a satellite connection from a
Linksys Router or PAP2 is getting a Not Acceptable Here (codec error) from
you can only make one 729 call at a time.
Jason Price
On 11/23/05, Aaron Clauson [EMAIL PROTECTED] wrote:
Hi,
I have a very strange Asterisk SIP call signalling
problem that is proving
extremely difficult to track down. The problem is that
any SIP INVITE
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Humberto Aicardi
Sent: 01 November 2005 17:17
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] PAP2 and ringing issues
Hi,
I currently have several PAP2-NA units configured to
Hi,
Is it possible to dynamically add contexts to the dial plan in any way?
Extensions can be added from the console and therefore also from MAPI but
their doesn't appear to be anyway to add a new context apart from reloading
the configuration files.
The reason I ask is my dialplan is getting
Hi,
If anyone has either:
- Found a company which ships these units outside the
US,
- Got one of the units and tried to unlock it from
Vonage.
Please post.
(The Linksys WRT54GP2 is the first acceptably priced
unit that has a router, WiFi and an ATA, at least that
I know of).
Aaron
Hi,
I haven't worked with Vonage myself but I usually get
this error back from my termination provider when the
number I have sent them is incorrect.
It might be worth checking you have used the correct
prefix (011 or 00) and area code etc.
Regards,
Aaron
Hi !
I have been working on
Hi,
I don't know if I missed something on the recent posts
regarding running * on the linksys boxes (couldn't
make any sense of the gifs that were posted??)?
Getting back to the original question, does anyone
know where the firmware or source for a linksys box
running * can be obtained?
Aaron
Hi,
Has anyone had any luck getting one of the new ZyXEL
P2602HW routers working with *??
These units look good on paper: DSL modem, 802.11g, 4
Port Ethernet, 2 x ATA plus all the bells and whistles
in the firmware.
It has 2 different SIP clients built in and I was able
to get them registered
Hi,
Thanks a lot for the configs Fabe.
I tried your zaptel.conf but I still get yellow and
red alarms in zttool and * is unable to create any Zap
channels (as expected with yellow and red alarms).
I realise I will now have to start talking to Colt (in
Ireland) to try and get the line up and
Hi,
Has anyone connected * to a Colt E1 line in Europe? If
so could you send me the zaptel.conf and zapata.conf.
Thanks,
Aaron
__
Do you Yahoo!?
New and Improved Yahoo! Mail - Send 10MB messages!
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[Kevin Walsh Wrote]
Marvellous. Microsoft will bring their legendary
stability, security
and reliability to the VoIP world.
Oops - there goes my lunch.
Maybe but looking past that what the unit will bring
is a programmable touch screen GUI on a hard VOIP
phone.
And being a Microsoft product
Hi,
I have started some users terminating calls from my
asterisk server to the PSTN through a couple of
termination providers.
The biggest problem I am having is the time it takes
to initially set the call up. It regularly exceeds
twenty seconds. I can work around this with failing
over to
Hi,
Found a nice little video about a prototype phone from
broadcom currently sitting in Microsoft WinCE lab. The
video is at:
http://channel9.msdn.com
The video in question is an interview with Mike Hall
titled Windows CE and Windows Embedded Lab Tour. The
clip dealing with the VOIP phone is
Hi,
I had a similar problem for a while in Ireland.
Eventually after much hair tearing I decided it must
be something to do with the phone socket and commenced
to make a direct conenction between the twisted pair
and the X100P socket. Low and behold it worked.
After more mucking around I found I
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Sandee
Sent: Wednesday, June 16, 2004 8:45 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cost of IP Phones, or
Isn't It Just
Software?
Am I dreaming?
Firstly
Hi,
Does anyone know of a list of internationally
accessible PSTN talking clocks?
I find talkjing clocks a good way to test the call
quality to a particular country.
There are a quite a few available in the US but the
only other two countries I have found numbers for are
the UK and Sweden.
Hi,
Yes I am contemplating writing yet another GUI
application for *. However I thought before I start
coding away I would see if anybody had any good ideas
about the interface. I have had a look around at the
other * GUIs and also a quick search of other PABX
GUIs but to my mind there was
Hi,
I am unable to get any music or sounds played with the
MusicOnHold or SayDigits commands. I do get sound from
the Playback and Background commands.
I have gone through the process of installing mpg123
and putting the link in usr/bin (and usr/local/bin).
For the MusicOnHold command I can see
Thanks for the suggestion about checking the wiring of
my telephone socket!
I was able to get my X100P to pass through the signal
and get rid of the Red Alarm in zttool, hallelujah!!!
My understanding of the problem was that the X100P
wants the POTS signal on pins 2 5 whereas the Irish
sockets
Ahhh this could be my problem! I just checked which
wires on the RJ11 cable had a voltage across them and
it was the yellow and green (3 4?). From what
someone posted the other day it's supposed to be
Bumble Bee and Christmas Tree.
I did have to get a technician out to fix my line when
it was
Hi,
Has anyone got the X100P to work with an anlogue line
in the Republic of Ireland?
I have the X100P installed but zttool indicates a Red
Alarm status on the card. It is on its own interrupt
and I have tried different PCI slots but all to no
avail.
Are there any alternatives to the X100P that
then the
phone line you have
plugged into the X100P is not working.
On Sat, 2004-05-15 at 03:17, Aaron Clauson wrote:
I have the X100P installed but zttool indicates a
Red
Alarm status on the card. It is on its own interrupt
and I have tried different PCI slots but all to no
avail
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