Thanks a lot. Works like a charm.
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On Tue, Dec 18, 2007 at 02:10:25PM +0100, Lars Bensmann wrote:
I will make some more tests and gather some CLI output.
han*CLI show hints
-= Registered Asterisk Dial Plan Hints =-
[EMAIL PROTECTED] : SIP/pioSIP/pio-mobi
State:IdleWatchers 2
srinivas Antarvedi wrote:
Hello all,
Here is the requirement from my side
to use Asterisk.NET API to generate
an automated call (outgoing) from asterisk
and then link to one of the extensions which
plays a sound file for the callee.
For this i have worked out in the follwing way
Good Day
Find attached the relevant portions of the asterisk CLI.
Please,which portion of the extension .conf should i send ?
It is connected via RJ 45 connector to an E1 modem to the telco company.
I use E1 link.
I will appreciate your reply.
Best Regards
On Dec 18, 2007 4:02 PM, dave
Thanks
Please am using putty to again access to my Linux asterisk box.
How can i use tcpdump to get your request on the exact Ethernet port and
port number.
I will appreciate your reply.
Best Regards
On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] wrote:
lolu,
sounds more like a
On Dec 20, 2007 12:33 PM, d tbsky [EMAIL PROTECTED] wrote:
Hi:
i am surveying ip phones for our company. we will use them with
asterisk.
we have office in taiwan, hong kong,singapore and china.
cisco and polycom are too expensive for us.
we try several china brand ip phones. they
Hi Steve
Am connected to the telco through an E1 link using modem(Watson 5 modem
SDHSL 1 PAIR schmid telecommunications).The MODEM is connected to the
asterisk box through RJ 45 to the asterisk box end and serial connector to
the modem end .
Which portion of the extension conf should i post ?
lolu,
while you are making the call., capture and post your CLI output
... this is easy to do since you are using putty.
login to your pbx and start asterisk, use the below command:
# asterisk -vvvr
then make the call. hilite the text on the putty terminal and paste it
into the body of the
What is the output of ztconfig from the Linux command line? What does
your zaptel.conf and zapata.conf look like? What is the relevant part
of extensions.conf (the dialout section that fails). Also from the CLI,
it would be most helpful to post the output you get when dialing out
fails. I
lol - yep when news of this first broke I thought thats actually a very
good idea to have implemented, though it sounds the way Trixbox
implemented it may have been unsecure.
Maybe someone else can come up with a better way of implementing this.
If the data was all randomised there's no harm in
Hello,
I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have
recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front
of the phone and also to hopefully resolve some issues with the phones
not registering after a long period.
Once we upgraded the phones now
hi:
my system has one 4-port fxo card and one 2-port E1 card.
for some reason, i like to place fxo as channel 1-4, and E1 use the
rest channels (5-66).
i modify zaptel.conf, and ztcfg -vv is happy. but asterisk seems
not happy with
this configuration. it still want channel 16 as D-channel, in
Chad,
You might want to upgrade to the latest firmware. I have 7961g on
8-3-3SR2S and works very well.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chad
Osmond
Sent: Thursday, December 20, 2007 10:33 AM
To: asterisk-users@lists.digium.com
Subject:
Hi all,
I am grateful for our contribution so far .
I followed dave advise and i have the attached file using the aterisk -r
when a call is made.
I attached two files.
One of the attached file is for the external call,which replied with the
PROBLEM all trunks are busy now,please try your
Hi All
I FOUND OUT THAT THE ATTACHMENT WAS NOT SENT WITH THE MAIL.
FIND BELOW THE OUTPUT USING asterisk -vvvr command for EXTERNAL calls that
gave the ouput ALL TRUNKS ARE BUSY PLEASE TRY YOUR CALL LATER.
Verbosity is at least 3
-- Executing Macro(SIP/7871-f813, dialout-trunk|1|018774957||)
hi gnubie:
snom seems has some re-brand ip phones. do they use the same firmware?
if they are the same, i don't understand why snom do this..
Regards,
tbskyd
2007/12/20, GNUbie [EMAIL PROTECTED]:
On Dec 20, 2007 12:33 PM, d tbsky [EMAIL PROTECTED] wrote:
Hi:
i am surveying ip phones
Hi guys,
I know that this could be considered a bit off the topic, I've just posted
this topic at VoIPSEC mailing list but I just thought this could be very
interesting for Asterisk community members so I'm posting it here too.
So the point is for traditional telephony we expect service
lolu
I reformated the output so it was easier to understand. I attached the
word document for you.
on the below line:
--
Executing Dial("SIP/7871-f813", "ZAP/1/8774957|120|W") in
new stack
-- Requested transfer capability:
0x00 - SPEECH
-- Called 1/8774957
-- Zap/1-1 is proceeding passing
Is it behind a router? either forward the necessary ports to the sip
phone's internal network ip address using the router, or move the phone
outside the router to get it an external network (global ip) ;)
Mojo
sandeep.s wrote:
Hi,
my sip phone is unreachable for external network(global ip)
Tilghman Lesher wrote:
On Wednesday 19 December 2007 17:44:15 shadowym wrote:
I had high hopes for this solution for unfortunately it's not working. Did
exactly as you specified but return path is still [EMAIL PROTECTED]
even though [EMAIL PROTECTED] in voicemail.conf :(
Did you
Tzafrir Cohen wrote:
On Tue, Dec 11, 2007 at 11:26:21PM +0800, Rilawich Ango wrote:
Hi,
How can I merge 2 gsm files into a single file? I have tried to use
soxmix as below but failed.
soxmix 1.gsm 2.gsm 1-2.gsm
The GNU coreutils are shipped with a special[1] tool for this task:
Hi,
I'm working on a 500 seats Asterisk project.
I'm wondering whether or not I should consider using Asterisk Realtime and a
database to manage phones registrations.
Stories in Dev mailing list say Realtime is mis-used or should be improved.
So, what's the bottom line ?
Can I consider anything
Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive.
We use them (SPA942) in our company. Everybody's happy.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
Hey, that works great!
Thanks!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC
Sent: Wednesday, December 19, 2007 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Thu, 20 Dec 2007, Mojo with Horan Company, LLC wrote:
Tzafrir Cohen wrote:
On Tue, Dec 11, 2007 at 11:26:21PM +0800, Rilawich Ango wrote:
Hi,
How can I merge 2 gsm files into a single file? I have tried to use
The GNU coreutils are shipped with a special[1] tool for this task:
cat
Hi All;
Is there any limitation on the number of users for
MeetMe Conference? In other words, how many parties
can join to the room and become a member of the room?
Is there any limitation?
Regards
Bilal
Hi List;
In the h323.conf file, the parameter gatekeeper is
used to let asterisk work as h323 gatekeeper listening
at port 1719 by setting gatekeeper=DISCOVER or it is
used to let asterisk search for the gatekeeper to talk
with it and receive calls from it? But if just to let
asterisk talk with
as far as I know it's unlimited and only tied to the capacity of a
single machines processing power.
Of course then all you need to do is tie multiple machines to the same
room in a daisy chain and expand from there.
Someone on the list a few months ago gave the example of chaining
together 5
We currently also use the Linksys SPA942 and SPA963 IP phones. They are very
nice phones, and very easy to manage.
Cheers,
Daniel Cole (CCNA)
Technical Support
Ph: 1800 424 683
Fax: 03 5221 7659
e: [EMAIL PROTECTED]
w: hugonet.com.au
Hi All;
I established h323 trunk using chan_h323 (between
asterisk and softswitch, already i did this using sip
and succeed, but now in h323 and i am facing a
problem).
The call reached to the softswitch, but it always
dropped when it send for destination, and we tried to
let softswitch to send
I use a TDM400 card to interface with my telco. I used asterisk voice
mail. However, if I'm on the telco line while another call comes in,
obviously it cannot go to asterisk voice mail but instead bounces to the
Telco voice mail.
Is there any means by which I can get asterisk detect the
On Thu, 2007-12-20 at 16:29 -0500, Jim Duda wrote:
I use a TDM400 card to interface with my telco. I used asterisk voice
mail. However, if I'm on the telco line while another call comes in,
obviously it cannot go to asterisk voice mail but instead bounces to the
Telco voice mail.
Is
I always forget about Linksys/Sipura phones. Yes they are very nice. I know
there can be some contractual issues when you deal with Linksys/Cisco, but
they are well constructed and I haven't heard anything bad.
I do think you'd get better support with Snom/Grandstream, but this is
coming from my
Daniel Cole wrote:
We currently also use the Linksys SPA942 and SPA963 IP phones. They are very
nice phones, and very easy to manage.
Both the 842 and 942 do not have lit displays and are hard to read.
Is the 963 any better?
Why anyone EVER built a LCD without backlighting is beyond me.
also, you want to think about transcoding... if you have different
technologies, the system load for transcoding would increase...
dean, cool, I didn't know you could hang a few * boxes together with
meetme...
daveC
Dean Collins wrote:
as far as I know it's unlimited and only tied to the
Brian J. Murrell wrote:
On Thu, 2007-12-20 at 16:29 -0500, Jim Duda wrote:
Is there any means by which I can get asterisk detect the Message
Waiting Indicator (MWI) from the telco? Is there some application or
variable which can be used to identify an active MWI from the Telco?
This
Hi,
We currently use the 942 and 962. These both have backlight displays. The 962
is obviously the model with the nice color screen.
These phones are very easy to manage. We configure all phone remotely via TFTP,
and they phones are set to pull their config periodically. Very easy for making
On Thu, 2007-12-20 at 17:17 -0500, John Novack wrote:
Both the 842 and 942 do not have lit displays and are hard to read.
Actually, the 942 *does* have a backlit LCD.
Is the 963 any better?
I'm assuming that's a typo and they really meant the 962. The 962 has a
bright color backlit LCD.
On Thu, 2007-12-20 at 23:39 +0800, d tbsky wrote:
hi:
my system has one 4-port fxo card and one 2-port E1 card.
for some reason, i like to place fxo as channel 1-4, and E1 use the
rest channels (5-66).
This will only work if you load the kernel driver for the fxo card
before the kernel
Chad,
I had the same problem when upgrading to some of the newer firmware. The newer
firmware gets even pickier (if that's even possible) about the config files. Go
the phone's webpage and look at the debug log. It will show you where it's not
parsing correctly. I'm not in front of my phone
My PHP script is using AMI's Originate command to make two-way calls.
The originate connects the first leg of the calls, plays a file to the
first called party, and then uses Dial() from the dialplan to dial the
other leg of the call.
I'm noticing that only about 30% of the calls make it through
I'm reminded of the (Pagoo??) call waiting voicemail applet a while back.
Let me sum up how one might do this: I'm using vitelity for an 800#
DID... it's $0.50/mo plus time used, which is around $0.02/min maybe.
get your asterisk box registered to your itsp so that when your 800# DID
is
Justin Killen wrote:
Hey, that works great!
Thanks!
-Original Message-
So, maybe place the phones in a context that waits for a four-digit id
_before_ matching it to the context you were initially trying:
Excellent :) No problem at all!
hi jsmith:
that explains everything. i didn't aware the module load sequence would
cause big difference. is there any document about this i am missing?
now the system is working as expected. i m glad that i asked and you
answered.
thanks a lot for your quick reply and help!!
Regards,
Dave,
I agree with you. I think it would be smarter to go to a new format how ever
one issues that a lot of people seem to have is when the syntax is changed.
This is why I suggested both. Maybe there can be a month (or maybe even two)
long discussion between the users and dev list for A)
Thank you very much Jared, this are good news!
Thanks again...
On Dec 19, 2007 7:03 PM, Jared Smith [EMAIL PROTECTED] wrote:
I just got off the phone with the software product manager who is over
AsteriskNOW, and have it on good authority that it will be released in
early January. There
Thanks Russell, that's what I'm looking for.
Any idea when this will become part an official asterisk release?
Jim
Russell Bryant wrote:
Brian J. Murrell wrote:
On Thu, 2007-12-20 at 16:29 -0500, Jim Duda wrote:
Is there any means by which I can get asterisk detect the Message
Waiting
I am using Realtime in virtually all my projects. So far, I haven't had
any major issues. It saves a lot of headache for profile/dialplan
updates, at least for me!
So I say, GO!
- Ben
Olivier wrote:
Hi,
I'm working on a 500 seats Asterisk project.
I'm wondering whether or not I should
20 dec 2007 kl. 01.43 skrev Dovid B:
snip
Our problem is that very few in the community test beta releases
or development code. I want to send a big thank you to all that do,
you are very important in this process. And for those of you who
want to join, go to www.asterisk.org and find
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