Hi All,
Please let me know the how to configure a SIP
trunk of a asterisk Server with another one (not IAX2).
Asterisk-A should register a SIP trunk with Asterisk-B
server .
With Regards
Alok Ranjan Mohapatra
Software Engineer
+91 9866269992
PrimeSoft IP Solutions
Hi,
I was wondering if anyone was aware of a work around/fix for the
following Polycom + Asterisk (1.2.7.1) bug, if this was doable in
Asterisk, or if it is just a bug in the Polycom firmware. (Appears in
both the 1.4.x release and the 1.6.7 releases)
The SIP message for voicemail notify on
Tzafrir Cohen ha scritto:
Works fine with Junghanns' cards.
One simple thing for you to test: set one port in TE mode and one port
in NT mode (move all 5 jumbers of that port to the other position to get
it into NT mode). Then try to make a loopback connection (using a
standard ethernet cable).
Andrew Joakimsen ha scritto:
Are you using WDS? While it won't totally fix every issue, I've found
in my trials that turning off WDS and making sure all the AP were
connected to the same wired network was way more reliable, no more
random unregistartion and issue with registering (still seems
Hello All,
I apologize beforehand if this is of topic. I am looking for
a solution for Wifi phones to access asterisk via
iax2 or sip. The trouble is that it is on a manufacturing shop floor, so
reception and noise cancellation is going to be a concern. Are there any non 2.4Ghz solutions
Hi,I have just installed fresh FreeBSD 6.1 and asterisk. But I get warning about power. I have read forums that others had the same problems, but there is no solution how to fix this problem. Maybe somebody knows how to fix this problem...
This is my machine configurations:KERNEL:options
On 27 Oct 2006, at 16:42, Roberto Pereyra wrote:
Hi
Which is most resistant to the loss of packages in a dirty link ?
SIP or IAX ?
Well there isn't much in it at the protocol level, at least in terms
of the
odd packet being lost. IAX does have an advantage when it comes to
the
More info.
[EMAIL PROTECTED] ~]# ztcfg -vv
Zaptel Configuration
==
Channel map:
0 channels configured.
cat /etc/modprobe.conf
alias scsi_hostadapter ahci
alias usb-controller ehci-hcd
alias usb-controller1 uhci-hcd
alias eth0 e100
alias
Tzafrir, please disregard my previous postdefinitely i was WAAAY sleepy.
The error was simple (now that I just waked up completely):
I was touching /etc/asterisk/zaptel.conf instead of
/etc/zaptel.conf. I should remind myself not to work past midnight.
I don't recall what made me think
Hi Friends,I have created SIP extensions in our two Asterisk servers. Now, I want to connect these two servers using SIP. I searched a lot in internet about this. But, I found that there is a possibility to connect two servers using IAX2 only. Is it possible to connect two Asterisk servers using
On Sat, Oct 28, 2006 at 09:24:52AM -0500, Erick Perez wrote:
More info.
[EMAIL PROTECTED] ~]# ztcfg -vv
Zaptel Configuration
==
Channel map:
ztcfg has heard of no channels, and hence did not try to configure them.
Why is that?
0 channels configured.
This is a problem of the codec you are attempting to use. Wich codec
is?, it seems to Asterisk cannot identify the codec you are using.
Regards
On 10/28/06, Giedrius Augys [EMAIL PROTECTED] wrote:
Hi,
I have just installed fresh FreeBSD 6.1 and asterisk. But I get warning
about power. I have
Hi List,
Im having a bit of an odd problem with asterisk and
outgoing zap calls.
Tzafrir has been kind enough to help me get the logging
sorted out so I have some idea of whats going wrong, but Im a
little flummoxed.
Essentially the symptoms are as follows;
Make a SIP call from
BTW: as an alternative to zaphfc+flotz, consider vzaphfc. It seems that
the only place from which you can download an up-to-date version
nowadays is the Debian zaptel package:
http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/
http://packages.debian.org/zaptel-source
Thanks! I
Alberto, you should have bought a dect solution, the dect technology is
far better at swapping between cells.
Wifi is still a little immature at this time.
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto
Pastore
Sent: Saturday, 28
On Sat, Oct 28, 2006 at 05:17:02PM +0200, Remco Barendse wrote:
BTW: as an alternative to zaphfc+flotz, consider vzaphfc. It seems that
the only place from which you can download an up-to-date version
nowadays is the Debian zaptel package:
Asteroid is a SIP denial of service attack tools which affected older versions
of Asterisk the Open Source PBX and may affect other products running the SIP
protocol. There are thousands of custom (mis)crafted SIP packets which were
sent to a older versions of Asterisk that caused errors stopping
On 2006-10-28 03:51:57 -0700, Alberto Pastore [EMAIL PROTECTED] said:
Andrew Joakimsen ha scritto:
Are you using WDS? While it won't totally fix every issue, I've found
in my trials that turning off WDS and making sure all the AP were
connected to the same wired network was way more reliable,
Steve,
I am trying to get tx_fax to work. I am using a TDM2401E card.
I have a 3 page fax and I only receive the first page on every attempt.
I think I have enabled debug output below.
Can you tell me what the problem might be?
I am using snapshot from oct 26. asterisk 1.2.13 and libtiff
On 2006-10-27 11:55:14 -0700, Andrew Joakimsen [EMAIL PROTECTED] said:
Are you using WDS? While it won't totally fix every issue, I've found in my
trials that turning off WDS and making sure all the AP were connected to the
same wired network was way more reliable, no more random
On 2006-10-28 07:55:43 -0700, Dean Collins [EMAIL PROTECTED] said:
Alberto, you should have bought a dect solution, the dect technology is
far better at swapping between cells.
Wifi is still a little immature at this time.
Not if correctly configured. This is simply wrong.
Marty
Hello,
I need to connect one diva server 4bri to a portuguese BRI interface.
The operator (PT) said that this bri is in point-to-multipoint mode
(S0). Previously one PBX has connected to that interface.
The asterisk and diva drivers are working ok but i cannot communicate
to outside via this
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Olivier wrote:
Hi,
It seems to me that Bristuff usage has reached a point which implies a
dedicated mailing list.
This list would be of major use for :
- bugs assessment
- features requests
- comments on Asterisk news
Who seconds that ?
On 20:46, Sat 28 Oct 06, Ron Wellsted wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Olivier wrote:
Hi,
It seems to me that Bristuff usage has reached a point which implies a
dedicated mailing list.
This list would be of major use for :
- bugs assessment
- features
It looks like this issue has been raised before, but I see it mostly
ignored and no answers given, so here it is again:
For the past couple of years, I've wanted a queue that works very
similarly to roundrobin/rrmemory, but that doesn't remember anything
about where the last ring went to. This
Hi,
I have an Asterisk behind NAT.
NAT=yes and canreinvite=no in globals and for the peer.
I call an peer. The peer advice to use another IP for the audio and my
Asterisk is sending audio stream to the Audio server.
Because of missing port forwarding I will not receive the audio stream and
hear
yup. use IAX
- Original Message -
From: Thomas Winter [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, October 29, 2006 1:26 AM
Subject: [asterisk-users] Asterisk behind NAT and without portforwarding
forrtp
Hi,
I have an Asterisk behind NAT.
NAT=yes and
Half asleep. Sorry for my last post. I believe you still need port
forwarding for IAX. Time to keep to my bed time.
- Original Message -
From: Thomas Winter [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, October 29, 2006 1:26 AM
Subject: [asterisk-users] Asterisk
Here's a weird problem that I'm not quite sure how to resolve. Zaptel
1.2.10 compiles just fine with make, but when make install is
run, this happens:
[ `id -u` = 0 ] /sbin/depmod -a 2.6.17-10-generic || :
[ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample
/etc/zaptel.conf
Im looking at setting up a VoIP GSM gateway to
connect to my asterisk box. What experience have people on this list have with GSM
gateway hardware. I have been looking at the 2N voiceblue products.
Steve
___
--Bandwidth and
I’m looking at setting up a VoIP GSM gateway to connect to my
asterisk box. What experience have people on this list have with
GSM gateway hardware. I have been looking at the 2N voiceblue
products.
We are using the voiceblue that supports a maximum of 4 x sims (and
are using all four
Peter,
How much does the 4 port cost? How many simultaneous calls can you make? Do
you need a mobile account from a mobile provider such as T-mobile?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter J Dean
Sent: Saturday, 28 October 2006 7:02 PM
To:
Hi,
I want to do so that extension 501 will always use trunk1 for outbound calls and 502 will use trunk2 for outboud calls. How do I do this. Right now all extensions use the same trunk for outbound calls.
-- Zeeshan A Zakaria
___
--Bandwidth and
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