Hi Neeraj,
my problem is not terminating but making the Cisco accept the calls
coming from my Asterisk. The bad news is I cannot have access to the
Cisco sw, it is like a black box for me. The only thing I can have
access to is the T1/E1 port on the back of the Cisco 2800.
I made a custom
- Original Message -
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, 2 July, 2010 5:08:14 PM
Subject: Re: [asterisk-users] Voiceprompts i.e. voicemail and conferencing in
multiple
Hello list,
what is the use of realtime SIP peers when you always need to reload the
sip configuration as if you were just putting your SIP peers in sip.conf ??
My SIP peers are now defined in a mysql-DB and when I add a mailbox in
the field 'mailbox', the change is not active untill a do a
On 06/07/10 10:34, Jonas Kellens wrote:
Hello list,
what is the use of realtime SIP peers when you always need to reload
the sip configuration as if you were just putting your SIP peers in
sip.conf ??
My SIP peers are now defined in a mysql-DB and when I add a mailbox in
the field
On 6 Jul 2010, at 10:34, Jonas Kellens wrote:
what is the use of realtime SIP peers when you always need to reload the sip
configuration as if you were just putting your SIP peers in sip.conf ??
Did you enable caching by any chance?
S
--
Hello,
this is my configuration :
;- REALTIME SUPPORT
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read realtime.txt and extconfig.txt in the /doc directory of the
; source code.
;
On 07/06/2010 12:00 PM, Ishfaq Malik wrote:
On 06/07/10 10:34, Jonas Kellens wrote:
Hello list,
what is the use of realtime SIP peers when you always need to reload
the sip configuration as if you were just putting your SIP peers in
sip.conf ??
My SIP peers are now defined in a mysql-DB
Hi,
When reading logs, I can see a couple of lines such as :
full.6:[Jun 30 15:53:26] NOTICE[6599] chan_dahdi.c: PRI got event: Alarm (4)
on Primary D-channel of span 1
full.6:[Jun 30 15:53:32] NOTICE[6599] chan_dahdi.c: PRI got event: No more
alarm (5) on Primary D-channel of span 1
full.6:[Jun
On 06/07/10 11:26, Jonas Kellens wrote:
On 07/06/2010 12:00 PM, Ishfaq Malik wrote:
On 06/07/10 10:34, Jonas Kellens wrote:
Hello list,
what is the use of realtime SIP peers when you always need to reload
the sip configuration as if you were just putting your SIP peers in
sip.conf ??
My
On Tue, Jul 6, 2010 at 1:09 AM, C.Savinovich
c.savinov...@itntelecom.com wrote:
I am writing to you privately because I am an asterisk consultant and if you
need any help I can help you for a fee.
Unfortunately your email is not private, now that it is on a public list.
--
Paul Belanger |
Hello Community,
I have a question , I have been working with asterisk and developed some
successful applications. I am facing an issue of security i.e. We deploy
servers to client end. Now i dont want the client to see my configuration
files (Of course copy and distribute or replicate the logic
On Tue, Jul 06, 2010 at 12:37:25PM +0200, Olivier wrote:
Hi,
When reading logs, I can see a couple of lines such as :
full.6:[Jun 30 15:53:26] NOTICE[6599] chan_dahdi.c: PRI got event: Alarm (4)
on Primary D-channel of span 1
full.6:[Jun 30 15:53:32] NOTICE[6599] chan_dahdi.c: PRI got
That's not right. Should be 1245 - 4512:
http://www.voip-info.org/wiki/view/crossover+T1+cable
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Tuesday, July 06, 2010 2:35 AM
To:
On Tue, Jul 6, 2010 at 7:40 AM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
Hello Community,
I have a question , I have been working with asterisk and developed some
successful applications. I am facing an issue of security i.e. We deploy
servers to client end. Now i dont want the
Thanks Bruce, I also think dahdi is not able to compile there, I see it
requires linux kernel.
I'll give a try with zaptel.. but...
Do you know if most up to date version of zaptel/solaris is at
solarisvoip.com?
Claudio
On Mon, 5 Jul 2010, Bruce McAlister wrote:
Hi Claudio,
As far as I am
On Tue, 6 Jul 2010, ABBAS SHAKEEL wrote:
Hello Community,
I have a question , I have been working with asterisk and developed some
successful applications. I am facing an issue of security i.e. We deploy
servers to client end. Now i dont want the client to see my configuration
files (Of
On Tue, Jun 29, 2010 at 10:39 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
I use SecureCRT+FX , and use ansi graphics.
Putty is nice w/WinSCP as well.
I'll +1 this - SecureCRT+FX is the first thing I got my employer to buy a
license of for me when I had to
Hi Peder,
I'make a new cable following the info on that webpage. I hope it works
with Cisco 2800 too! :)
Thank you!
Giorgio Incantalupo
Peder wrote:
That's not right. Should be 1245 - 4512:
http://www.voip-info.org/wiki/view/crossover+T1+cable
-Original Message-
From:
How can I match any_num_of_digits#any_num_of_digits in an IVR?
I want users to be able to type, eg., 123#4567
I tried the following but it hangs up immediately. If I uncomment WaitExten
then it hangs up right when the user dials #.
As a side question, can I play a background message while using
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Tuesday, July 06, 2010 8:11 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] IVR
How can I match any_num_of_digits#any_num_of_digits
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can set
Hi,
How do I specify to which parking lot the hints refer to?
For exemple, on 1.4 I use this to see whether a call is parked in 800:
exten = 800,hint,park:8...@parkedcalls
But on 1.6 I have multiple parking lots working apparently sucessfully. How
do I build the hint for
On Tue, 6 Jul 2010, C.Savinovich wrote:
I am writing to you privately... [snip]
Doh! Need another cup of coffee?
j
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New to Asterisk? Join us for a
On Tue, Jul 6, 2010 at 10:19 AM, unsero...@aol.com wrote:
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In
2010/7/6 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Jul 06, 2010 at 12:37:25PM +0200, Olivier wrote:
Hi,
When reading logs, I can see a couple of lines such as :
full.6:[Jun 30 15:53:26] NOTICE[6599] chan_dahdi.c: PRI got event: Alarm
(4)
on Primary D-channel of span 1
Hi,
I'll soon replace a Junghanns OctoBRI with a Junghanns QuadBRI.
As both use wcb4xxp driver (dahdi 2.3.0, libpri 1.4.10.2 and asterisk
1.6.1), I'm planning to proceed this way :
1. Edit 2 versions of files /etc/dahdi/system.conf and
/etc/asterisk/dahdi-channels.conf (one for each card).
2.
On 07/05/2010 09:02 AM, Kristijan Vrban wrote:
Hello, i just had some fax abortions because of some packet loss. so i
startet to examine in the pcap recording
from the res_fax_digium, if the T.38 EC mode redundancy was really
used. So i watched into it, and compared it
with a t.38 pcap from
Just downloaded PrivateSHELL and it seems to be what everyone is looking for
in Putty. It's much better than putty in terms of not being sluggish and
scrolling is fine. Plus the window and the text doesn't hurt your eyes. It
has One click SFTP as well. So, good bye to WinSCP.
I think I found what
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the dialplan you can
Hi,
I'm having problems with a TE420P card, in which I cannot make calls
using spans 2 through 4.
After a couple of days of working correctly, spans 2, 3 and 4 start
failing (can not make calls). The system is configured to work with SS7.
After the ACM message goes out, immediately a REL
On Tue, Jul 06, 2010 at 05:03:04PM +0200, Olivier wrote:
Hi,
I'll soon replace a Junghanns OctoBRI with a Junghanns QuadBRI.
As both use wcb4xxp driver (dahdi 2.3.0, libpri 1.4.10.2 and asterisk
1.6.1), I'm planning to proceed this way :
1. Edit 2 versions of files /etc/dahdi/system.conf
Hello every one:
I have instaled asterisk in a open suse 10.2 operative system and i try to
probe two softphone inside my LAN but i alway receive the same error call
failed:error 408 timeout and i don`t have any error in a /var/log/asterisk and
any in the CLI i hope your can help me
Bye and
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.
In you are just using Asterisk in the
- Original Message -
- Original Message -
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
that
we are unable to URI dial our clients.
Good Afternoon,
Can someone please explain what Y-cords are available out there and how they
can be used with Aastra or other VoIP phones? Maybe with or WITHOUT
headsets?
Isn't a Y-cord traded for soft Barge in these days?
Thanks,
Bruce
--
Tuesday, July 6, 2010, 8:11:46 PM, bruce wrote:
Can someone please explain what Y-cords are available out there and how they
can be used with Aastra or other VoIP phones? Maybe with or WITHOUT
headsets?
Isn't a Y-cord traded for soft Barge in these days?
I think Y-cords only for PSTN. Or
I believe people use this for headsets, to have a superviror listen in on a
call with the agent (for training purposes). You can therefore plug in two
headsets on the same phone.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
SIP user = Asterisk 1.6 server = SIP Trunk = external destination:
works
AMI script = Asterisk 1.6 server = SIP Trunk = external destination:
Failed to authenticate on INVITE to 'asterisk
sip:asterisk@(ipaddr);tag=alphanumeric'
I¹ve tried doing things like ³include = contextwithtrunk in various
We deal with Y-cords all the time for Ethernet and BRIs. They are just
normal cords, making use of the fact that both Cat5 networks and BRI ports
don't use all the 8 pins, so why not use extra wires in the cable for
something useful instead of wasting them. It has nothing to do with the
Hi
In sip.conf, you generally have something like
[name]
..
username=
secret=
What is the difference between the name specified in brackets and the
username key ?
What the sip client should provide ?
What do we use in dialplan when trying to reach this client ?
--
On Tue, Jul 6, 2010 at 4:10 PM, Mike Ely mike...@amyskitchen.net wrote:
Obviously, I'm playing around with the context a bit but for now just want
to get the outbound call working.
debug log would be helpful:
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
--
On Tue, Jul 6, 2010 at 7:11 PM, Ruddy Gbaguidi plugwo...@micnes.com wrote:
What is the difference between the name specified in brackets and the
username key ?
Context and username.
What the sip client should provide ?
The client will tell you their settings
What do we use in dialplan when
Log attached. It looks like the call is trying to do an invite to the sip
trunk and fails there - it never actually tries to send the destination to
the ShoreTel system on the other end of the trunk. Here's the ShoreTel
context from sip.conf:
[ShoreTel]
type=peer
qualify=yes
port=5060
On Tue, Jul 6, 2010 at 8:00 PM, Mike Ely mike...@amyskitchen.net wrote:
Log attached.
--- SIP read from UDP:10.10.10.16:5060 ---
SIP/2.0 401 Unauthorized
context from sip.conf:
[ShoreTel]
type=peer
qualify=yes
port=5060
host=10.10.10.16
context=incoming
canreinvite=no
Your context is
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of Paul Belanger
Sent: Tue 7/6/2010 5:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject:Re: [asterisk-users] Can't dial out through AMI
On Tue, Jul 6, 2010 at 8:00
Not if a voicemail is left by another system, but if a voicemail is deleted by
an external system (ie. web interface). But externnotify is only run upon
voicemail() or voicemailmain()? What is the purpose of pollmailboxes=yes then?
-Eric
From: tles...@digium.com
To:
Hi,
How do I configure Asterisk as a Video Conference purpose. What package
I need to configure and what steps I need to follow to configure in
dial-plan to authenticate user.
Regards,
Hiren Mistry
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