Hi,
There is someone (Tony?) with disconnection problems (after about 15s) when
calling between two DIAX phones? I have a voicemessage regarding this issue,
without any contact address.
If yes, please send me more details about configuration (iax.conf and
extensions.conf files, IAX mode, etc.).
Some of you have been following our progress on
http://farfon.convergence.com.pk as we blundered our way through the
development of a low-cost ethernet IP phone that does IAX and augments the
client options currently available for the kick-assterisk server.
With help from the denizens of
Hi all,
I have installed FAX app as described in several mails.
When a fax call is received, I get the following in the * console:
-- Starting simple switch on 'Zap/1-1'
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Ringing(Zap/1-1, ) in new stack
-- Executing
Hi,
I have started the * server in console mode (-vc) and this is what I get
now (no file saved and disconnected):
*CLI
-- Starting simple switch on 'Zap/1-1'
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Ringing(Zap/1-1, ) in new stack
-- Executing Wait(Zap/1-1,
Hi Dan,
Dan wrote:
Hi,
I have started the * server in console mode (-vc) and this is what I get
now (no file saved and disconnected):
[]
It seems the software FAX modem is sending out its messages regularly,
but never hears anything recognisable come back from the far end FAX
Hi guys,
I am in the process of establishing a Call Center. I need some suggestions from those who have already worked on such setups using Asterisk. My scenerio is:
US T1 --- Asterisk gw 1 -[gsm compression]-- Asterisk gw 2 [with TDM10B] --- Sip Phones [Xlite]
Calls from US are landing
Hi Steve,
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 07, 2003 10:36 AM
Subject: Re: [Asterisk-Users] RxFAX application
Hi Dan,
Dan wrote:
Hi,
I have started the * server in console mode (-vc) and this is what I
Dan wrote:
Hi Steve,
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 07, 2003 10:36 AM
Subject: Re: [Asterisk-Users] RxFAX application
Hi Dan,
Dan wrote:
Hi,
I have started the * server in console mode (-vc) and
Hi,
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 07, 2003 11:11 AM
Subject: Re: [Asterisk-Users] RxFAX application
It seems the software FAX modem is sending out its messages regularly,
but never hears anything recognisable
On Sat, 2003-12-06 at 17:55, Cameron Jacobson wrote:
I have just started laying out the plans for my first project using
Asterisk. I am very interested at this stage in getting much needed
feedback, critiquing my approach. What are the ups and downs going to
be if I develop this project
Its the VT1000
http://broadband.motorola.com/catalog/productdetail.asp?ProductID=212
We have looked everywhere for it but looks like no distributor sells it
right now.
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 07, 2003 4:50
--On Sunday, December 07, 2003 09:36:14 -0500 TeleSIP [EMAIL PROTECTED]
wrote:
Its the VT1000
http://broadband.motorola.com/catalog/productdetail.asp?ProductID=212
We have looked everywhere for it but looks like no distributor sells it
right now.
Maybe because it's a new variant of the
hi,
i am getting make update error in asterisk directory
=
it is possible that you first compiled asterisk and
unrecognized request ' then zaptel thus'
cvs update: dying gasp from cvs.digium.com unexpected
I for one would be very interested in seeing your moded makefiles, I'm
also trying to use 2.6 wherever possible.
--
Dave Cotton [EMAIL PROTECTED]
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[EMAIL PROTECTED]
same here :)
On Sunday 07 of December 2003 16:23, Dave Cotton wrote:
I for one would be very interested in seeing your moded makefiles, I'm
also trying to use 2.6 wherever possible.
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[EMAIL PROTECTED]
What kind of stability / reliability are people currently experiencing
with the Linux / Asterisk combination? We will be running 3-10 SIP
phones from India to US using nothing more than regular cable / dsl
connections from both locations.
People have had months of uptime. I would be more
-Original Message-
From: Walker Haddock
Sent: Thursday, December 04, 2003 7:54 PM
To: [EMAIL PROTECTED]
We have an installation with 9 inbound voice channels (one is
the fax) and 768K data. It is a Hybrid PRI. It terminates
into a T100P. It is working great! The cost was
Philipp von Klitzing wrote:
For H323 you'll need to install a gatekeeper next to Asterisk and fiddle
with h323 or oh323 (I love to live dangerously, hit me Jeremy). :-
Moreover NetMeeting doesn't work through NAT.
A gatekeeper is TOTALLY optional in an H.323 network. Read the spec.
Jeremy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Thursday, December 04, 2003 8:48 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Channelbank Recomendation and
GS102 question
At 8:15 PM -0500 12/4/03, Jim Flagg wrote:
oops, my apologiee for attaching the entire source of the drivers :( Here's a
diff
Index: wcfxo.c
===
RCS file: /usr/cvsroot/zaptel/wcfxo.c,v
retrieving revision 1.21
diff -a -u -r1.21 wcfxo.c
--- wcfxo.c 17 Nov 2003
FYI,
the usual place for patch's is on
bugs.digium.com
and then a little link in an email here to let ppl know about it ..
This allows a single place where these get reviewed for inclusion in cvs
..nice work
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On Sat, 06 Dec 2003 10:10:21 -0700, Michael Welter wrote
Hi,
I have a RH9 system with an onboard VIA sound chip.
According to the archives, VIA won't work for asterisk.
So, I disabled the VIA and I purchased a Creative Labs
Soundblaster PCI 128-Voice soundcard ($13). This card is
on
It's not really intended to be applied at this stage since it will break 2.4.
I am intending to update the bug in digiums database with my findings once I get
the chance.
On Sun, Dec 07, 2003 at 09:59:16AM -0800, TC wrote:
FYI,
the usual place for patch's is on
bugs.digium.com
and then a
Thanks guys,
I found a SoundBlaster 16 PCI at a different CompUSA store, and
everything is working perfect.
Grzegorz Nosek wrote:
On Sat, 06 Dec 2003 10:10:21 -0700, Michael Welter wrote
Hi,
I have a RH9 system with an onboard VIA sound chip.
According to the archives, VIA won't work for
Would have probably been more appropriate to at least announce that
iax was going to disappear at some specific date, as opposed to folks
randomly discoverying it and chasing problems. (Kind of related to why
there isn't a marketing plan.)
Sorry, it was something of a side effect of
Good, glad to hear things are better. Without getting into too much techie
detail, what was the root problem?
There was just race that I introduced a while back. If calls came in
while a reload was taking place in IAX2, bad things would happen. Now
it's fixed. Originally I was thinking it
On Sunday 07 of December 2003 21:14, Mark Spencer wrote:
Good, glad to hear things are better. Without getting into too much
techie detail, what was the root problem?
There was just race that I introduced a while back. If calls came in
while a reload was taking place in IAX2, bad things
Are you guys using power over Ethernet?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Sunday, December 07, 2003 2:32 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] FARFON lives!
Some of you have been
On Wed, 2003-12-03 at 15:34, William Waites wrote:
localnet= internal ip of * machine?
localnet should be the internal network address not the internal
ip address. i.e. if your asterisk server is 192.168.0.245, localnet
should be 192.168.0.0
Agreed, I was wrong before :)
--
Leif Madsen
We are using an MGCP configuration. There seems to
be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is
how our Vendor sees it:
Here's what I see.1. The first call is initiated.
(CRCX) The interesting thing here is that the CA (Call Agent) tells us to
go
I'm trying to find this 'posting'. For some reason I'm missing it. Can
anyone point it out please?
- the host= setting (plus deny=/permit=) in particular is
what can create
the unexpected headaches if used with type=friend (some weeks
ago there
was an excellent posting on this issue,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: Thursday, December 04, 2003 3:48 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102
question
Very interesting. I've had now two fights with
I've been using NuFone with Asterisk for a while, but I've started
seeing this error with incoming calls:
NOTICE[114696]: File chan_iax2.c, Line 4581 (socket_read): Rejected
connect attempt from 216.234.116.189, requested/capability 0x4/0x4
incompatible with our capability 0xff03.
Outgoing
I'm trying to come up with an elegant solution to handle roaming users in a
branch office scenario. I have a number of possible scenarios, none of
which seem to completely solve the problem. Perhaps someone with a better
feel of the interactions can help me out. Is the 'switch' statement useful
Hello all,
I am newbie to Telephony world (IP and PSTN). Please excuse me if you find my
questions very dumb.
I am trying to configure my IP 7940 with the Asterisk, when phone boots up it only
shows the message Phone Unprovisioned on the LCD panel.
Under Settings--SIP Configuration--Line 1
On Fri, 2003-12-05 at 01:11, Jonathan Tew wrote:
Being farely new to the Asterisk scene and searching for
documentation I was wondering why the asterisk.org site didn't run a
wiki. There isn't anything as good as a wiki for collecting
collaborative documentation. Over time someone
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 4:59 AM
Subject: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included
Hi all,
A new version (0.9.6) of DIAX is available for download at:
Any ideas on how to do this one?
FWD requires an * on certain calls as a prefix character, but I cannot
seem to be able to get Prefix(*) to add that to the front of the
extension that is dialed... Setting up an extension that dials
(SIP/[EMAIL PROTECTED]) works just fine, but in trying to add
On Sunday 07 December 2003 22:48, Kris Stark wrote:
Any ideas on how to do this one?
FWD requires an * on certain calls as a prefix character, but I
cannot seem to be able to get Prefix(*) to add that to the front of
the extension that is dialed... Setting up an extension that dials
exten = _7X,2,Dial(SIP/*${EXTEN:[EMAIL PROTECTED])
On Sun, 7 Dec 2003, Kris Stark wrote:
Any ideas on how to do this one?
FWD requires an * on certain calls as a prefix character, but I cannot
seem to be able to get Prefix(*) to add that to the front of the
extension that is dialed...
On Sun, 2003-12-07 at 23:57, Tilghman Lesher wrote:
On Sunday 07 December 2003 22:48, Kris Stark wrote:
Any ideas on how to do this one?
FWD requires an * on certain calls as a prefix character, but I
cannot seem to be able to get Prefix(*) to add that to the front of
the extension that
I have three quad FXS cards (in slots 1-3) and one quad FXO card in slot
4. I have a few telephone sets connected to the FXS cards, and * allows
calling from one phone to another and from phone to console, etc. Diax
also works.
I have a CO line connected to one circuit of the the FXO card.
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